Avaya Aura™ Communication Manager Overview

Avaya Aura™ Communication Manager Overview
Avaya Aura™ Communication
Manager Overview
03-300468
Issue 6
Release 5.2
May 2009
© 2009 Avaya Inc.
All Rights Reserved.
Notice
While reasonable efforts were made to ensure that the information in this
document was complete and accurate at the time of printing, Avaya Inc. can
assume no liability for any errors. Changes and corrections to the information
in this document may be incorporated in future releases.
For full legal page information, please see the complete document,
Avaya Legal Page for Software Documentation.
To locate this document on the website, simply go to
http://www.avaya.com/support and search for the document number in the
search box.
Documentation disclaimer
Avaya Inc. is not responsible for any modifications, additions, or deletions to
the original published version of this documentation unless such modifications,
additions, or deletions were performed by Avaya. Customer and/or End User
agree to indemnify and hold harmless Avaya, Avaya's agents, servants and
employees against all claims, lawsuits, demands and judgments arising out of,
or in connection with, subsequent modifications, additions or deletions to this
documentation to the extent made by the Customer or End User.
Link disclaimer
Avaya Inc. is not responsible for the contents or reliability of any linked Web
sites referenced elsewhere within this documentation, and Avaya does not
necessarily endorse the products, services, or information described or offered
within them. We cannot guarantee that these links will work all of the time and
we have no control over the availability of the linked pages.
Warranty
Avaya Inc. provides a limited warranty on this product. Refer to your sales
agreement to establish the terms of the limited warranty. In addition, Avaya’s
standard warranty language, as well as information regarding support for this
product, while under warranty, is available through the following Web site:
http://www.avaya.com/support
Copyright
Except where expressly stated otherwise, the Product is protected by copyright
and other laws respecting proprietary rights. Unauthorized reproduction,
transfer, and or use can be a criminal, as well as a civil, offense under the
applicable law.
Avaya support
Avaya provides a telephone number for you to use to report problems or to ask
questions about your product. The support telephone number
is 1-800-242-2121 in the United States. For additional support telephone
numbers, see the Avaya Web site:
http://www.avaya.com/support
Contents
Chapter 1: Communication Manager Overview . . . . . . . . . . . . . .
25
Optional software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
26
Capacities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
26
Avaya Installation Wizard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
26
Gateway Installation Wizard . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
28
System Management Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . .
28
Avaya 8XXX Servers and media gateways . . . . . . . . . . . . . . . . . . . . . .
Multi-Tech gateway support . . . . . . . . . . . . . . . . . . . . . . . . . . .
Co-residency of Communication Manager
and SIP Enablement Services . . . . . . . . . . . . . . . . . . . . . . . . . .
29
29
Chapter 2: Application programming interface . . . . . . . . . . . . . .
31
Application Enablement Services . .
Software-only option . . . . . . .
Bundled server option . . . . . .
CVLAN . . . . . . . . . . . . . . .
Web services . . . . . . . . . . .
Telephony Service . . . . . .
System Management Service .
User Service . . . . . . . . . .
.
.
.
.
.
.
.
.
31
31
32
32
32
32
32
33
Device and media control API . . . . . . . . . . . . . . . . . . . . . . . . . . . .
33
DEFINITY LAN Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Adjunct switch application interface . . . . . . . . . . . . . . . . . . . . . . .
34
34
JTAPI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
34
TSAPI
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
34
Chapter 3: Attendant features . . . . . . . . . . . . . . . . . . . . . . .
37
Accessing the attendant . . . .
Dial access to attendant. . .
Individual attendant access .
Recall . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
37
37
37
37
Attendant backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
38
Attendant room status. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
38
Attendant functions using Distributed Communications System protocol
Control of trunk group access . . . . . . . . . . . . . . . . . . . . . .
Direct trunk group selection . . . . . . . . . . . . . . . . . . . . . . .
Inter-PBX attendant calls . . . . . . . . . . . . . . . . . . . . . . . . .
.
.
.
.
38
38
39
39
Call handling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Attendant Intrusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
39
39
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
29
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
3
Contents
Attendant lockout - privacy . . . . . . . . .
Attendant split swap. . . . . . . . . . . . .
Attendant vectoring . . . . . . . . . . . . .
Automated attendant . . . . . . . . . . . .
Backup alerting . . . . . . . . . . . . . . .
Call waiting . . . . . . . . . . . . . . . . . .
Calling of inward restricted stations . . . .
Conference . . . . . . . . . . . . . . . . . .
Enhanced Return Call to (same) Attendant
Listed directory number. . . . . . . . . . .
Override of diversion features . . . . . . .
Priority queue . . . . . . . . . . . . . . . .
Release loop operation . . . . . . . . . . .
Selective conference mute . . . . . . . . .
Serial calling . . . . . . . . . . . . . . . . .
Timed reminder and attendant timers . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
39
39
40
40
40
40
40
41
41
41
41
42
42
42
42
42
Centralized Attendant Service . . . . . . . . . . . . . . . . . . . . . . . . . . . .
43
Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
43
Increased attendant consoles . . . . . . . . . . . . . . . . . . . . . . . . . . . .
43
Making calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Auto Start and Do Not Split . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Auto Manual Splitting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
44
44
44
Monitoring calls . . . . . . . . . . . . . . . . . . . . .
Attendant control of trunk group access . . . . .
Attendant direct extension selection . . . . . . . .
Attendant direct trunk group selection. . . . . . .
Crisis alerts to an attendant console . . . . . . . .
Trunk group busy/warning indicators to attendant
Trunk identification by attendant . . . . . . . . . .
Visually Impaired Attendant Service . . . . . . . .
.
.
.
.
.
.
.
.
44
44
45
45
45
46
46
46
Chapter 4: Call Center . . . . . . . . . . . . . . . . . . . . . . . . . . .
47
Computer Telephony Integration . . . . . . . . . . . .
Adjunct route support for network call redirection
Co-resident DEFINITY LAN Gateway . . . . . . . .
Direct Agent Announcement . . . . . . . . . . . .
Flexible billing . . . . . . . . . . . . . . . . . . . .
Pending work mode change . . . . . . . . . . . .
Trunk group identification . . . . . . . . . . . . .
User-to-User Information propagation during
4 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
47
47
48
48
48
49
49
Contents
manual transfer/conference operations. . . . . . . . . . . . . . . . . . . . .
Block CMS Move Agent events . . . . . . . . . . . . . . . . . . . . . . . . .
VDN override for ASAI messages . . . . . . . . . . . . . . . . . . . . . . . .
49
49
50
Automatic Call Distribution . . . . . . . . . . .
Abandoned Call Search . . . . . . . . . . .
Interruptible Aux work. . . . . . . . . . . .
Adjunct Routing . . . . . . . . . . . . . . .
Auto-Available Split . . . . . . . . . . . . .
Automatic Number Identification . . . . .
Incoming Automatic Number Identification
Outgoing Automatic Number Identification
Local feedback for queued ACD calls . . .
Queue status indicators . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
50
51
52
52
52
52
52
53
53
54
Avaya Basic Call Management System . . . .
Avaya Business Advocate . . . . . . . . .
Auto reserve agents . . . . . . . . . . .
Call selection override per skill . . . . .
Dynamic percentage adjustment . . . .
Dynamic queue position. . . . . . . . .
Dynamic threshold adjustment . . . . .
Logged-in advocate agent counting . .
Percent allocation distribution . . . . .
Reserve agent time in queue activation
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
54
55
55
55
55
55
55
55
56
56
Avaya Call Center features supported on the Avaya G700 Media Gateway . . . .
56
Avaya Call Management System . . . . . . . . . . . . . . . . . . . . . . . . . . .
Avaya virtual routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enhanced information forwarding . . . . . . . . . . . . . . . . . . . . . . . .
57
57
57
Call center release control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
58
Call prompting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Data collection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Data In/Voice Answer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
58
58
58
Call vectoring . . . . . . . . . . . . . . . . . . . . . . . . . . .
Advanced vector routing . . . . . . . . . . . . . . . . . . .
Average Speed of Answer routing . . . . . . . . . . . .
Best service routing . . . . . . . . . . . . . . . . . . . .
Best service routing polling over IP without B-channel.
Expected Wait Time routing. . . . . . . . . . . . . . . .
Call center messaging. . . . . . . . . . . . . . . . . . . . .
Percentage allocation routing . . . . . . . . . . . . . . . .
Holiday vectoring . . . . . . . . . . . . . . . . . . . . . . .
59
59
59
59
59
60
60
60
60
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
5
Contents
Vector Directory Number . . . . .
Class of Restriction for VDN .
Display VDN for route-to DAC.
VDN in a coverage path . . . .
VDN of origin announcement .
VDN return destination . . . .
.
.
.
.
.
.
60
61
61
61
61
61
Call Work Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
62
Caller Information Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . .
62
Circular station hunt group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
62
Clear the display of collected digits . . . . . . . . . . . . . . . . . . . . . . . . .
62
CMS measurement of ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
63
Dialed Number Identification Service . . . . . . . . . . . . . . . . . . . . . . . .
63
Direct agent calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
63
Dual links to CMS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
63
Duplicate agent login ID administration . . . . . . . . . . . . . . . . . . . . . . .
Agent-loginID skill pair increase . . . . . . . . . . . . . . . . . . . . . . . . .
64
64
Expert Agent Selection . . . . . .
Add/remove skills . . . . . . .
Call distribution based on skill
Queue to best ISDN support .
.
.
.
.
64
64
65
65
Least Occupied Agent . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
65
Multiple call handling (forced) . . . . . . . . . . . . . . . . . . . . . . . . . . . .
65
Multiple music/audio sources. . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Locally sourced announcements and music . . . . . . . . . . . . . . . . . .
66
66
Multiple split queuing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
66
Network Call Redirection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
ETSI Explicit Call Transfer signaling . . . . . . . . . . . . . . . . . . . . . . .
Network call redirection 2B-channel transfer . . . . . . . . . . . . . . . . . .
66
67
67
PC Application Software Translation Exchange . . . . . . . . . . . . . . . . . . .
67
Priority queuing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
67
Reason codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
68
Redirection on no answer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
68
Remote logout of agent . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
68
Service observing . . . . . . . . . . . . . .
Listen-only FAC for service observing
Service observing by COR . . . . . . .
Service observing of VDNs . . . . . . .
Service observing remote . . . . . . . .
68
69
69
69
69
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
6 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Service Observing with Multiple Observers . . . . . . . . . . . . . . . . . . .
Vector-initiated service observing . . . . . . . . . . . . . . . . . . . . . . . .
69
70
Site statistics for remote port networks . . . . . . . . . . . . . . . . . . . . . . .
70
User-to-user information over the public network . . . . . . . . . . . . . . . . .
70
Voice Response Integration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
71
VuStats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
71
Chapter 5: Collaboration . . . . . . . . . . . . . . . . . . . . . . . . . .
73
Conferencing . . . . . . . . . . . . . . . . . . . . . . .
Abort conference on hang-up. . . . . . . . . . . . .
Conference - three party . . . . . . . . . . . . . . .
Conference - six party . . . . . . . . . . . . . . . . .
Conference/transfer display prompts . . . . . . . .
Conference/transfer toggle/swap . . . . . . . . . . .
Group listen . . . . . . . . . . . . . . . . . . . . . .
Hold/unhold conference. . . . . . . . . . . . . . . .
Meet-me Conferencing . . . . . . . . . . . . . . . .
Expanded Meet-me Conferencing . . . . . . . . . .
No dial tone conferencing. . . . . . . . . . . . . . .
No hold conference . . . . . . . . . . . . . . . . . .
Select line appearance conferencing. . . . . . . . .
Selective conference party display, drop, and mute
Selective conference mute . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
73
73
73
73
74
74
74
74
75
75
75
75
76
76
77
Multimedia calling . . . . . . . . . . . . . . . . . . . . . . .
Multimedia Application Server Interface . . . . . . . . .
Multimedia call early answer on vectors and stations .
Multimedia Call Handling . . . . . . . . . . . . . . . . .
Multimedia call redirection to multimedia endpoint . . .
Multimedia data conferencing (T.120) through an ESM .
Multimedia hold, conference, transfer, and drop . . . .
Multimedia queuing with voice announcement . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
77
78
78
78
79
79
79
79
Paging and intercom . . . . . .
Code calling access . . . . .
Group paging . . . . . . . .
Intercom - automatic. . . . .
Intercom - automatic answer
Intercom - dial . . . . . . . .
Loudspeaker paging access
Manual signaling. . . . . . .
Whisper page . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
79
79
80
80
80
80
81
81
81
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
7
Contents
Chapter 6: Communication device support . . . . . . . . . . . . . . . .
83
Avaya IP Agent . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
83
Avaya IP Softphone . . . . . . . . . . . . . . . . . . .
IP Softphone and IP Agent - RoadWarrior mode .
IP Softphone and IP Agent - Shared Control mode
IP Softphone and IP Agent - Telecommuter mode
.
.
.
.
83
84
84
84
Avaya IP Softphone for pocket PC . . . . . . . . . . . . . . . . . . . . . . . . . .
84
Communication Manager PC console . . . . . . . . . . . . . . . . . . . . . . . .
85
Avaya one-X Communicator . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
85
Avaya one-X Portal as software-only phone . . . . . . . . . . . . . . . . . . . . .
85
Avaya SIP softphone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
86
Avaya SoftConsole . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Avaya SoftConsole - RoadWarrior mode . . . . . . . . . . . . . . . . . . . .
Avaya SoftConsole - Telecommuter mode . . . . . . . . . . . . . . . . . . .
86
86
87
Increased text field length for feature buttons - DCP . . . . . . . . . . . . . . . .
87
Unicode support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
QSIG support for Unicode . . . . . . . . . . . . . . . . . . . . . . . . . . . .
87
88
Chapter 7: Hospitality . . . . . . . . . . . . . . . . . . . . . . . . . . . .
89
Alphanumeric dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
89
Attendant room status. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
89
Automatic selection of Direct Inward Dialing numbers . . . . . . . . . . . . . . .
89
Automatic wakeup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
89
Check-in/check-out . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
90
Custom selection of VIP DID numbers . . . . . . . . . . . . . . . . . . . . . . . .
90
Daily wakeup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
90
Dial-by-name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
90
Do not disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
91
Dual wakeup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
91
Housekeeping status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
91
Names registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
91
Property Management System digit to insert/delete . . . . . . . . . . . . . . . .
92
Property Management System interface . . . . . . . . . . . . . . . . . . . . . . .
92
Single-digit dialing and mixed station numbering. . . . . . . . . . . . . . . . . .
92
Suite check-in . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
93
VIP wakeup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
93
Wake-up activation using confirmation tones . . . . . . . . . . . . . . . . . . . .
93
8 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Xiox call accounting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
93
Chapter 8: Localization . . . . . . . . . . . . . . . . . . . . . . . . . . .
95
Administrable language displays. . . . . . . . . . . . . . . . . . . . . . . . . . .
95
Administrable loss plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
95
Bellcore calling name ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
95
Block collect call. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
96
Busy tone disconnect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
96
Country-specific localization . . . . . . . . . . . . . . .
Italy . . . . . . . . . . . . . . . . . . . . . . . . . . .
Distributed Communications Systems protocol.
Japan . . . . . . . . . . . . . . . . . . . . . . . . . .
National private networking support . . . . . . .
Katakana character set . . . . . . . . . . . . . .
Russia . . . . . . . . . . . . . . . . . . . . . . . . .
Central Office support on G700 Media Gateway .
ISDN/DATS network support . . . . . . . . . . .
Multi-Frequency Packet signaling . . . . . . . .
.
.
.
.
.
.
.
.
.
.
96
96
96
97
97
97
97
97
97
97
E&M signaling - continuous and pulsed . . . . . . . . . . . . . . . . . . . . . . .
98
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Multinational Locations . . . . . . . . . . . . . . . . . . . . . . . . .
Analog line board parameters per location . . . . . . . . . . . .
Companding for DCP telephones and circuit packs per location
Location ID in Call Detail Record records . . . . . . . . . . . . .
Loss plans per location . . . . . . . . . . . . . . . . . . . . . . .
Multifrequency signaling per trunk group . . . . . . . . . . . . .
Tone generation per location . . . . . . . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
98
99
99
99
100
100
100
Public network call priority . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
100
World class tone detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
XOIP Tone Detection Bypass . . . . . . . . . . . . . . . . . . . . . . . . . . .
101
101
Chapter 9: Message integration . . . . . . . . . . . . . . . . . . . . . .
103
™
Avaya Aura Communication Manager Messaging . . . . . . . . . . . . . . . .
CM Messaging application . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Record on messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
103
103
106
Audible message waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
107
Leave Word Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Leave Word Calling - QSIG/DCS . . . . . . . . . . . . . . . . . . . . . . . . .
107
108
Manual message waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Centralized voice mail (Tenovis) . . . . . . . . . . . . . . . . . . . . . . . . .
108
108
Issue 6 May 2009
9
Contents
Message retrieval . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
109
Message Sequence Tracer enhancements. . . . . . . . . . . . . . . . . . . . . .
109
Octel integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
109
QSIG/DCS voice mail interworking . . . . . . . . . . . . . . . . . . . . . . . . . .
Multiple QSIG voice mail hunt groups . . . . . . . . . . . . . . . . . . . . . .
110
110
Voice mail retrieval button . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
110
Voice message retrieval . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
110
Voice messaging and call coverage . . . . . . . . . . . . . . . . . . . . . . . . .
111
Chapter 10: Mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
113
Administration Without Hardware . . . . . . . . . . . . . . . . . . . . . . . . . .
113
Automatic Customer Telephone Rearrangement . . . . . . . . . . . . . . . . . .
113
Avaya Wireless Telephone Solutions. . . . . . . . . . . . . . . . . . . . . . . . .
114
Avaya Extension to Cellular . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Off-PBX station . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
114
115
E911 ELIN for IP wired extensions . . . . . . . . . . . . . . . . . . . . . . . . . .
E911 device location for IP telephones . . . . . . . . . . . . . . . . . . . . .
116
117
Personal Station Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Do not answer reason code . . . . . . . . . . . . . . . . . . . . . . . . . . .
Name/number permanent display . . . . . . . . . . . . . . . . . . . . . . . .
117
117
118
Seamless Converged Communication Across Networks . . . . . . . . . . . . . .
118
SIP Visiting User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
118
Terminal Translation Initialization . . . . . . . . . . . . . . . . . . . . . . . . . .
119
TransTalk 9000 digital wireless system . . . . . . . . . . . . . . . . . . . . . . .
119
X-station mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
119
Chapter 11: Port network and gateway connectivity . . . . . . . . . . .
121
Asynchronous Transfer Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Port Network Connectivity . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Port Network Connectivity over WAN . . . . . . . . . . . . . . . . . . . . . .
121
121
121
Circuit switched . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Center Stage Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Center Stage Switch separation . . . . . . . . . . . . . . . . . . . . . . . . .
122
122
122
Internet Protocol . . . . . . . . . . .
H.248 media gateway control . .
Inter-Gateway Alternate Routing
Network Region Wizard . . .
IP Port Network Connectivity . .
122
122
123
123
123
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
10 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Improved Port Network Recovery from Control Network Outages . . . . . . .
Link Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
123
124
Separation of Bearer and Signaling . . . . . . . . . . . . . . . . . . . . . . . . .
125
Chapter 12: Trunk connectivity . . . . . . . . . . . . . . . . . . . . . .
127
Asynchronous Transfer Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Circuit Emulation Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
CMS measurement of ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . .
127
127
127
Circuit switched . . . . . . . . . . . . . . . . . . . . . . .
DS1 trunk service . . . . . . . . . . . . . . . . . . . .
Echo cancellation - with UDS1 circuit pack . . . .
E1 . . . . . . . . . . . . . . . . . . . . . . . . . . .
T1 . . . . . . . . . . . . . . . . . . . . . . . . . . .
Separate licensing for TDM stations and TDM trunks
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
127
127
128
128
128
128
Internet Protocol . . . . . . . . .
H.323 trunk . . . . . . . . .
Improved button downloads
Increased trunk capacity . .
IP loss groups . . . . . . . .
IP trunks . . . . . . . . . . .
IP trunk fallback to PSTN . .
IP trunk link bounce . . . . .
Session Initiation Protocol .
SIP trunks . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
129
129
129
129
130
130
131
131
131
132
Auxiliary trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Advanced Private Line Termination . . . . . . . . . . . . . . . . . . . . . . .
132
132
Central Office . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Central Office support on G700 Media Gateway - Russia . . . . . . . . . . .
133
133
Digital multiplexed interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Bit-oriented signalling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Message-oriented signalling . . . . . . . . . . . . . . . . . . . . . . . . . . .
133
133
133
Direct Inward Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
134
Direct Inward/Outward Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . .
134
E&M signaling - continuous and pulsed . . . . . . . . . . . . . . . . . . . . . . .
134
E911 CAMA trunk group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
134
Foreign Exchange . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
134
ISDN trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Automatic Termination Endpoint Identifier . . . . . . . . . . . . . . . . . . .
Call-by-call service selection . . . . . . . . . . . . . . . . . . . . . . . . . . .
135
135
135
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
11
Contents
ETSI functionality . . . . . . . . . . . . . . .
ETSI completion of calls . . . . . . . . .
Facility and non-facility associated signaling
Feature plus . . . . . . . . . . . . . . . . . .
ISDN-Basic Rate Interface. . . . . . . . . . .
Multiple subscriber number - limited . . . . .
NT interface on TN556C . . . . . . . . . . . .
Presentation restriction . . . . . . . . . . . .
Wideband switching . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
135
136
136
136
136
138
139
139
139
Multi-Frequency Packet signaling - Russia . . . . . . . . . . . . . . . . . . . . .
139
National private networking support - Japan . . . . . . . . . . . . . . . . . . . .
139
Personal Central Office Line . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
140
Release Link Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
140
Remote access trunks. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
140
Tie trunks. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
140
Timed automatic disconnect for outgoing trunk calls . . . . . . . . . . . . . . .
140
Wide Area Telecommunications Service . . . . . . . . . . . . . . . . . . . . . .
141
Chapter 13: Public Networking and connectivity . . . . . . . . . . . . .
143
Caller ID on analog trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
143
Caller ID on digital trunks
143
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
DS1 trunk service . . . . . . . . . . . . . . . .
Echo cancellation - with UDS1 circuit pack
E1 . . . . . . . . . . . . . . . . . . . . . . .
T1 . . . . . . . . . . . . . . . . . . . . . . .
.
.
.
.
143
143
143
144
Flexible billing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
144
Local exchange trunks . . . . . . . . . . . .
800-service trunks . . . . . . . . . . . . .
Central Office trunks . . . . . . . . . . .
Digital Service 1 trunks . . . . . . . . . .
Direct Inward Dialing trunks . . . . . . .
Direct Inward/Outward Dialing trunks . .
Foreign Exchange trunks . . . . . . . . .
Wide Area Telecommunications Service
.
.
.
.
.
.
.
.
144
144
144
144
144
145
145
145
. . . . . . . . . . . . . . . . .
145
Chapter 14: Intelligent networking . . . . . . . . . . . . . . . . . . . . .
147
Avaya VoIP Monitoring Manager . . . . . . . . . . . . . . . . . . . . . . . . . . .
147
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
QSIG Supplementary Service - Advice of Charge
12 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Distributed Communications System protocol
Attendant with DCS . . . . . . . . . . . . .
Direct trunk group selection . . . . . .
Display . . . . . . . . . . . . . . . . . .
DCS automatic circuit assurance . . . . . .
DCS over ISDN-PRI D-channel . . . . . . .
DCS protocol - Italy . . . . . . . . . . . . .
DCS with reroute. . . . . . . . . . . . . . .
QSIG/DCS voice mail interworking . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
147
147
147
147
147
148
148
148
148
Electronic Tandem Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Automatic alternate conditional routing . . . . . . . . . . . . . . . . . . . . .
Trunk signaling and error recovery . . . . . . . . . . . . . . . . . . . . . . .
148
148
149
Extension number portability . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
149
Internet Protocol . . . . . . . . . . . . . . . . . . . . . . . . .
Alternate gatekeeper and registration addresses . . . . .
Classless Interdomain Routing . . . . . . . . . . . . . .
Multiple network regions per CLAN . . . . . . . . . . . .
Multiple location support for network regions . . . . . .
Daylight Savings Time rules change . . . . . . . . . .
Network regions . . . . . . . . . . . . . . . . . . . . . . .
Processor Ethernet . . . . . . . . . . . . . . . . . . . . .
Adjuncts . . . . . . . . . . . . . . . . . . . . . . . . .
Merge of IP Connect and Multiconnect configurations
H.248 and H.323 registration . . . . . . . . . . . . . .
S8500 Servers . . . . . . . . . . . . . . . . . . . . . .
Quality of Service . . . . . . . . . . . . . . . . . . . . . .
802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . .
Camp-on/Busy-out . . . . . . . . . . . . . . . . . . . .
Call Admission Control bandwidth management . . .
CLAN load balancing . . . . . . . . . . . . . . . . . .
Codecs . . . . . . . . . . . . . . . . . . . . . . . . . .
Differentiated services. . . . . . . . . . . . . . . . . .
Dynamic jitter buffers . . . . . . . . . . . . . . . . . .
Integration with Cajun rules. . . . . . . . . . . . . . .
IP overload control. . . . . . . . . . . . . . . . . . . .
IPSI administration enhancements . . . . . . . . . . .
QoS for call control . . . . . . . . . . . . . . . . . . .
QoS for VoIP . . . . . . . . . . . . . . . . . . . . . . .
QoS to endpoints . . . . . . . . . . . . . . . . . . . .
Resource Reservation Protocol . . . . . . . . . . . .
149
150
151
151
151
151
151
152
152
153
153
154
154
154
154
155
155
155
155
155
155
156
156
156
156
157
157
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
13
Contents
Sending and receiving faxes over IP . .
Modem over IP . . . . . . . . . . . . .
Relay mode . . . . . . . . . . . . . . .
Pass through mode . . . . . . . . . .
Encryption . . . . . . . . . . . . . . .
T.38 faxes over the Internet . . . . . . .
Pass through mode . . . . . . . . . .
Shuffling and hairpinning . . . . . . . .
G.722 shuffling over H.323/SIP trunks
NAT with shuffling . . . . . . . . . . .
TTY . . . . . . . . . . . . . . . . . . . . .
TTY over analog and digital trunks . .
TTY over Avaya IP trunks . . . . . . .
TTY relay mode . . . . . . . . . . . .
TTY pass through mode . . . . . . .
Variable length ping . . . . . . . . . . . .
Variable Length Subnet Mask . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
157
158
158
158
159
159
160
160
160
160
161
162
162
163
163
163
164
QSIG . . . . . . . . . . . . . . . . . . . . . . . . . . .
Auto callback - QSIG Call Completion . . . . . . .
Basic . . . . . . . . . . . . . . . . . . . . . . . . .
Call completion . . . . . . . . . . . . . . . . . . .
Call forwarding (diversion) . . . . . . . . . . . . .
Call Independent Signaling Connections . . . . .
Call offer . . . . . . . . . . . . . . . . . . . . . . .
Call transfer . . . . . . . . . . . . . . . . . . . . .
Name display on unsupervised transfer . . . .
Called name ID . . . . . . . . . . . . . . . . . . . .
Centralized Attendant Service . . . . . . . . . . .
Attendant display of Class of Restriction . . .
Attendant return call . . . . . . . . . . . . . . .
Priority queue . . . . . . . . . . . . . . . . . .
RLT emulation through a PRI . . . . . . . . . .
Communication Manager/Octel QSIG integration .
Complex private numbering plan support . . . . .
Leave Word Calling . . . . . . . . . . . . . . . . .
Manufacturer-Specific Information . . . . . . . .
Message Waiting Indication . . . . . . . . . . . . .
Name and number identification . . . . . . . . . .
Path replacement with path retention . . . . . . .
QSIG/DCS voice mail interworking . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
164
164
164
164
165
166
166
166
166
166
167
167
167
167
167
167
167
168
168
168
168
169
169
14 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Reroute after diversion to voice mail. .
Stand-alone path replacement . . . . .
Supplementary services and rerouting
VALU . . . . . . . . . . . . . . . . . . .
Call coverage. . . . . . . . . . . . .
Call coverage and CAS . . . . . . .
Distinctive alerting . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
169
169
170
170
170
170
170
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
171
171
171
172
173
Chapter 15: Data interfaces . . . . . . . . . . . . . . . . . . . . . . . .
175
Administered connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
175
Data call setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
175
Data hot line . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
175
Data privacy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
175
Data restriction. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
176
Default dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
176
IP asynchronous links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
176
Multimedia application server interface . . . . . . . . . . . . . . . . . . . . . . .
177
Multimedia calling . . . . . . . . . . . . . . . . . . . . . .
Multimedia call early answer on vectors and stations
Multimedia Call Handling . . . . . . . . . . . . . . . .
Multimedia call redirection to multimedia endpoint . .
Multimedia data conferencing (T.120) through ESM .
Multimedia hold, conference, transfer, and drop . . .
Multimedia multiple-port networks . . . . . . . . . . .
.
.
.
.
.
.
.
177
177
178
179
179
179
179
Pass advice of charge information to world class BRI endpoints . . . . . . . . .
179
Chapter 16: Call routing . . . . . . . . . . . . . . . . . . . . . . . . . .
181
Alternate facility restriction levels . . . . . . . . . . . . . . . . . . . . . . . . . .
181
Automatic routing features . . . . . .
Automatic Alternate Routing . . .
Automatic Route Selection . . . .
ARS/AAR dialing without FAC
AAR/ARS overlap sending . .
AAR/ARS partitioning . . . . .
181
181
182
182
182
182
Uniform Dial Plan . . . . . . . . . .
Dial Plan Expansion . . . . . . .
Multi-location dial plans . . . . .
Punctuation on station displays
Extended trunk access . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
15
Contents
AAR/ARS partitioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
183
Enbloc Dialing and Call Type Digit Analysis . . . . . . . . . . . . . . . . . . . . .
183
Generalized route selection .
Look-ahead routing . . .
Node number routing . .
Time of day routing . . .
.
.
.
.
183
184
184
184
Multiple location support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Multiple location support for network regions . . . . . . . . . . . . . . . . .
184
185
Traveling class marks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
185
Answer detection . . . . . . . . . .
Answer supervision by time-out
Call-classifier board . . . . . . .
Network answer supervision . .
.
.
.
.
185
185
185
185
Chapter 17: Reliability and survivability . . . . . . . . . . . . . . . . . .
187
Alternate gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
187
Auto fallback to primary for H.248 gateways . . . . . . . . . . . . . . . . . . . .
187
Connection preserving failover/failback for H.248 media gateways . . . . . . . .
188
Connection preserving upgrades for duplex servers
. . . . . . . . . . . . . . .
188
Enterprise Survivable Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Automatic return to primary server. . . . . . . . . . . . . . . . . . . . . . . .
Dial Plan Transparency for LSP and ESS . . . . . . . . . . . . . . . . . . . .
189
189
189
IP bearer duplication using the TN2602AP circuit pack . . . . . . .
Load balancing . . . . . . . . . . . . . . . . . . . . . . . . . . .
Bearer signal duplication . . . . . . . . . . . . . . . . . . . . . .
Reduced channels with duplicated TN2602AP circuit packs
.
.
.
.
190
190
191
191
IP endpoint Time-to-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
192
Local Survivable Processor . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
192
Handling of split registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . .
193
Multiple network regions per CLAN . . . . . . . . . . . . . . . . . . . . . . . . .
193
Power failure transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
193
Standard Local Survivability . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
194
Survivable Remote EPN
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
194
Chapter 18: Security, privacy, and safety . . . . . . . . . . . . . . . . .
197
System administrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Authentication, Authorization, and Accounting Services . . . . . . . . . . .
Access security gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
197
197
197
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
16 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Branch gateway enhancements . . . . . . . . . . . . . . .
Alternate facility restriction levels . . . . . . . . . . . . . . . .
Alternate operations support system alarm number . . . . . .
Privacy - attendant lockout . . . . . . . . . . . . . . . . . .
Authorization codes - 13 digits . . . . . . . . . . . . . . . . . .
Call restrictions . . . . . . . . . . . . . . . . . . . . . . . . . .
Class of Restriction . . . . . . . . . . . . . . . . . . . . . . . .
Block collect call . . . . . . . . . . . . . . . . . . . . . . . .
Customer-provided equipment alarm . . . . . . . . . . . . . .
Data privacy . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Data restriction. . . . . . . . . . . . . . . . . . . . . . . . . . .
Encryption algorithm for bearer channels . . . . . . . . . . .
SRTP media encryption . . . . . . . . . . . . . . . . . . . .
Enhanced security logging . . . . . . . . . . . . . . . . . . . .
Facility restriction levels and traveling class marks . . . . . .
H.248 link encryption . . . . . . . . . . . . . . . . . . . . . . .
Malicious call trace . . . . . . . . . . . . . . . . . . . . . . . .
Malicious call trace logging . . . . . . . . . . . . . . . . .
Mask station name and number for internal calls . . . . . . .
Media encryption . . . . . . . . . . . . . . . . . . . . . . . . .
License file requirements . . . . . . . . . . . . . . . . . . .
PIN Checking for Private Calls . . . . . . . . . . . . . . . . . .
Restriction - controlled . . . . . . . . . . . . . . . . . . . . . .
Secure shell and secure FTP . . . . . . . . . . . . . . . . . . .
Security of IP telephone config files . . . . . . . . . . . . . . .
Security of IP telephone registration/H.323 signaling channel
Security Violation Notification . . . . . . . . . . . . . . . . . .
Signaling encryption for SIP trunks . . . . . . . . . . . . . . .
Station security codes . . . . . . . . . . . . . . . . . . . . . .
Tripwire security . . . . . . . . . . . . . . . . . . . . . . . . . .
End user . . . . . . . . . . . . . . . . . . . . . .
Backup alerting . . . . . . . . . . . . . . . .
Barrier codes. . . . . . . . . . . . . . . . . .
Calling/Connected Party Number restriction
Per call CPN restriction . . . . . . . . . .
Per line CPN restriction . . . . . . . . . .
Crisis alerts to a digital numeric pager. . . .
Crisis alerts to a digital station . . . . . . . .
Crisis alerts to an attendant console . . . . .
Emergency access to the attendant . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
197
198
198
198
198
199
199
199
199
199
200
200
200
201
201
201
202
202
202
202
203
204
204
205
205
205
206
206
206
206
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
207
207
207
207
207
208
208
208
209
209
Issue 6 May 2009
17
Contents
E911 CAMA trunk group . .
Hot Desking Enhancement .
Privacy - auto exclusion. . .
Privacy - manual exclusion .
Restriction - controlled . . .
Station lock . . . . . . . . .
Station lock by Time of Day
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
209
209
210
210
210
210
211
Chapter 19: System management . . . . . . . . . . . . . . . . . . . . .
213
Administration change notification . . . . . . . . . . . . . . . . . . . . . . . . .
213
Administration Without Hardware . . . . . . . . . . . . . . . . . . . . . . . . . .
213
Alternate facility restriction levels . . . . . . . . . . . . . . . . . . . . . . . . . .
213
Announcements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
214
Authorization codes - 13 digits . . . . . . . . . . . . . . . . . . . . . . . . . . . .
214
Automatic circuit assurance . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
214
Automatic transmission measurement system . . . . . . . . . . . . . . . . . . .
214
Avaya Directory Enabled Management . . . . . . . . . . . . . . . . . . . . . . .
215
Avaya Integrated Management . . . . . . . . . . . . . . .
Communication Manager configuration manager . . .
Communication Manager fault/performance manager
Avaya site administration . . . . . . . . . . . . . . . .
Voice Announcement over LAN manager . . . . . . .
Increased announcement support . . . . . . . . .
Avaya VoIP Monitoring Manager . . . . . . . . . . . .
Lightweight Directory Access Protocol . . . . . . . .
.
.
.
.
.
.
.
.
215
215
216
216
216
216
217
217
Barrier codes. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
217
Bulletin board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
217
Busy verification of telephones and trunks . . . . . . . . . . . . . . . . . . . . .
218
Call charge information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
218
Call Detail Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Call Detail Recording display of physical extension . . . . . . . . . . . . . .
Legacy CDR and Survivable CDR . . . . . . . . . . . . . . . . . . . . . . . .
219
219
219
Call restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
220
Calling party/billing number . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
220
Class of Restriction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
220
Class of Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
220
Classless Interdomain Routing
. . . . . . . . . . . . . . . . . . . . . . . . . . .
221
Concurrent user sessions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
222
18 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Customer-provided equipment alarm . . . . . . . . . . . . . . . . . . . . . . . .
222
Customer telephone activation
. . . . . . . . . . . . . . . . . . . . . . . . . . .
222
DCS automatic circuit assurance . . . . . . . . . . . . . . . . . . . . . . . . . .
222
External device alarming . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
223
Facility busy indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
223
Facility restriction levels and traveling class marks . . . . . . . . . . . . . . . .
223
Facility test calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
223
Firmware download . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
224
Five EPN maximum in MCC1 Media Gateways . . . . . . . . . . . . . . . . . . .
224
Information and reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enhanced logging of user actions . . . . . . . . . . . . . . . . . . . . . . . .
Parsing capabilities for the history report . . . . . . . . . . . . . . . . . . . .
225
227
227
IP asynchronous links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
228
Malicious call trace . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Malicious call trace logging . . . . . . . . . . . . . . . . . . . . . . . . . . .
228
228
Music-on-hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Local music-on-hold. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Multiple music sources . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
228
229
229
Restriction - controlled . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
229
Scheduling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
230
Security Violation Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . .
230
Station security codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
230
Tenant partitioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
230
Terminal Translation Initialization . . . . . . . . . . . . . . . . . . . . . . . . . .
231
Time of day clock synchronization through a LAN source . . . . . . . . . . . . .
Linux platforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
UNIX platforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
231
231
231
Trunk group circuits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
231
Variable length ping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
232
Variable Length Subnet Mask . . . . . . . . . . . . . . . . . . . . . . . . . . . .
232
Chapter 20: Telecommuting and remote office . . . . . . . . . . . . . .
233
Coverage of calls redirected off-net . . . . . . . . . . . . . . . . . . . . . . . . .
233
Extended user administration of redirected calls (telecommuting access) . . . .
233
IP Softphone and IP Agent - RoadWarrior mode . . . . . . . . . . . . . . . . . .
233
IP Softphone and IP Agent - Shared Control mode . . . . . . . . . . . . . . . . .
234
IP Softphone and IP Agent - Telecommuter mode. . . . . . . . . . . . . . . . . .
234
Issue 6 May 2009
19
Contents
IP Softphone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
234
Off-premises station . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
234
Remote access. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
234
Chapter 21: Telephony . . . . . . . . . . . . . . . . . . . . . . . . . . .
235
Abbreviated Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Abbreviated dialing labeling . . . . . . . . . . . . . . . . . . . . . . . . . . .
Abbreviated dialing on-hook programming . . . . . . . . . . . . . . . . . . .
235
235
235
ABCD tone support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
236
Active dialing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
236
Administrable timeout on call timer . . . . . . . . . . . . . . . . . . . . . . . . .
236
Alphanumeric dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
236
Automatic Call Back . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Automatic Call Back for analog telephones . . . . . . . . . . . . . . . . . . .
237
237
Automatic hold. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
237
Avaya video telephony solution . . . . . . . . . . . . . . . . . . . . . . . . . . .
237
Bellcore calling name ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
238
Bridged call appearance - multi-appearance telephone . . . . . . . . . . . . . .
238
Bridged call appearance - single-line telephone . . . . . . . . . . . . . . . . . .
239
Call coverage. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Alphanumeric field designation . . . . . . . . . . . . . . . . . . . . .
Changeable coverage paths . . . . . . . . . . . . . . . . . . . . . . .
Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enhanced coverage and ringback for logged off IP/PSA/TTI stations
Time of day . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
239
239
239
240
240
240
Call redirection . . . . . . . . . . . . . . . . . .
Call forward busy/do not answer . . . . . . .
Call forwarding all calls . . . . . . . . . . . .
Call forwarding enhancements . . . . . .
Chained call forwarding . . . . . . . . . . . .
Enhanced Redirection Notification . . . . . .
Call forwarding override. . . . . . . . . . . .
Send All Calls and Call Forwarding Override
Call redirection intervals . . . . . . . . . . .
Call Log enhancements . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
240
240
241
241
241
241
241
242
242
242
Call park . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
242
Call pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enhanced Call Pickup Alerting . . . . . . . . . . . . . . . . . . . . . . . . . .
243
243
20 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Group call pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
243
Caller ID on analog trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
243
Caller ID on digital trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
243
Circular station hunt group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
243
Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
244
Consult . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
244
Coverage callback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
244
Coverage incoming call identification . . . . . . . . . . . . . . . . . . . . . . . .
244
Disconnecting unanswered calls . . . . . . . . . . . . . . . . . . . . . . . . . . .
244
Distinctive ringing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Maintain external ring tone after internal transfer . . . . . . . . . . . . . . .
245
245
Edit dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
245
Emergency calls from unnamed IP endpoints
. . . . . . . . . . . . . . . . . . .
245
Enhanced abbreviated dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . .
246
Enhanced telephone display . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
246
Enterprise Mobility User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enterprise Mobility User enhancements . . . . . . . . . . . . . . . . . . . .
247
247
Enterprise Wide Licensing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
248
Go to cover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
248
Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
248
Intercom - automatic answer . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
248
Internal automatic answer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
248
Last number dialed . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
249
Local call timer automatic start/stop . . . . . . . . . . . . . . . . . . . . . . . . .
249
Long hold recall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
249
Manual originating line service . . . . . . . . . . . . . . . . . . . . . . . . . . . .
249
Misoperation handling. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
250
Multiappearance preselection and preference. . . . . . . . . . . . . . . . . . . .
250
Multiple level precedence and preemption
Announcements for precedence calling
Dual homing . . . . . . . . . . . . . . .
End office access line hunting . . . . .
Line load control . . . . . . . . . . . . .
Precedence call waiting . . . . . . . . .
Precedence calling . . . . . . . . . . .
Precedence routing . . . . . . . . . . .
Preemption . . . . . . . . . . . . . . . .
251
251
252
252
252
252
252
253
253
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Issue 6 May 2009
21
Contents
Worldwide numbering and dialing plan . . . . . . . . . . . . . . . . . . . . .
254
Night service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Enhanced night service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
254
254
License modes . . . . . . . . . . . . . .
License-normal mode . . . . . . . . .
License-error mode . . . . . . . . . .
Limit the number of concurrent calls
No-license mode . . . . . . . . . . .
.
.
.
.
.
255
255
255
256
256
Personalized ringing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
257
Posted messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
257
Priority calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
257
Pull transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
258
Recall signaling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
258
Recorded telephone dictation access . . . . . . . . . . . . . . . . . . . . . . . .
258
Reset shift call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
258
Ringback queuing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
259
Ringer cutoff . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
259
Ringing - abbreviated and delayed . . . . . . . . . . . . . . . . . . . . . . . . . .
259
Ringing options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
259
Send all calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
259
Special dial tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
260
Station hunting. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
260
Station hunt before coverage . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
260
Station self display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
260
Station used as a virtual extension . . . . . . . . . . . . . . . . . . . . . . . . . .
261
Support for the Hewlett Packard DL380G2 server. . . . . . . . . . . . . . . . . .
261
Team button . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
261
Telephone display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
ISO 8859-1 encoding compatibility . . . . . . . . . . . . . . . . . . . . . . .
262
262
Telephone self administration . . . . . . . . . . . . . . . . . . . . . . . . . . . .
263
Temporary bridged appearance . . . . . . . . . . . . . . . . . . . . . . . . . . .
263
Terminating extension group . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
263
Time of day routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
264
Timed call disconnection for outgoing trunk calls . . . . . . . . . . . . . . . . .
264
Transfer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Abort transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Transfer - outgoing trunk to outgoing trunk . . . . . . . . . . . . . . . . . . .
264
265
265
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
22 Avaya AuraTM Communication Manager Overview
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
Transfer recall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Transfer upon hang-up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Trunk-to-trunk transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
265
265
265
Trunk flash . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
266
Index
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
267
Issue 6 May 2009
23
Contents
24 Avaya AuraTM Communication Manager Overview
Chapter 1: Communication Manager Overview
Avaya Aura™ Communication Manager organizes and routes voice, data, image and video
transmissions. It can connect to private and public telephone networks, ethernet LANs, ATM
networks, and the Internet.
Communication Manager seeks to solve business challenges by powering voice
communications and integrating with value-added applications. Communication Manager is an
open, scalable, highly reliable and secure telephony application. Communication Manager
provides user and system management functionality, intelligent call routing, application
integration and extensibility, and enterprise communications networking Figure 1: System
running Communication Manager on page 25.
Figure 1: System running Communication Manager
1
2
3
4
cydfswtc KLC 030102
Figure notes:
1.
Voice
2.
Data
3.
Image
4.
Multimedia
Issue 6 May 2009
25
Communication Manager Overview
Optional software
Various optional packages can enhance the capabilities of your system. Some of the
capabilities described in this document require optional software. See your Avaya account
representative for more information.
Capacities
System capacities have been expanded for many products and features. However, the most
up-to-date system capacity information is not listed in Communication Manager documentation.
For the entire list of updated capacities, see Avaya Aura™ Communication Manager System
Capacities Table, 03-300511.
To view the system capacity limits table,
1. Go to the Avaya Web site.
2. Locate the latest version of the system capacities table document, and then click the title of
the document to download the information.
Avaya Installation Wizard
The Avaya Installation Wizard is a tool to be used in installations (not upgrades) of
Communication Manager in certain system configurations. The Installation Wizard helps with
reduced complexity, time-to-install, and the cost of installation.
The Avaya Installation Wizard does not work with all media gateways and Avaya 8XXX Servers.
For more information, contact your Avaya representative.
! CAUTION:
The Avaya Installation Wizard and the web installation that is accessible from the
System Management Interface should not be run at the same time. Make sure
you complete one process before you start the other process.
CAUTION:
The Installation Wizard delivers the following installation advantages:
l
Intuitive user interface with on-line help
l
Auto-discovery, where appropriate
l
No assumption of external internet connectivity
l
Ease of updating to newest software & firmware
26 Avaya AuraTM Communication Manager Overview
Avaya Installation Wizard
l
Ability to import customized name & number list
l
Complete record of all settings
l
Guided process from beginning to end
The Installation Wizard can guide installers through:
l
License file and authentication file setup
l
Avaya 8XXX Server & media gateway configuration
l
Telephony, trunk, and endpoint configuration and installation
l
Installation log file summary creation
The Installation Wizard for Communication Manager has these features:
l
Support for SAGE 16xx IP Phones
l
Support for 9670G IP Phone (Large Screen Model)
l
Configure S8400-Series Server as an ESS
l
Configure CM Messaging (formerly called IA770) on S8500-Series Servers
l
Configure S85xx and S84xx-Series Servers for Remote Maintenance Board second
ethernet interface
l
Configure S87xx-Series Servers using encrypted software-based duplication
l
Configure Memory (Standard or Extra Large)
l
The Installation Wizard supports a stack of up to 10 G700 Media Gateways.
l
l
l
l
l
l
l
l
Technicians are able to load updated media module firmware versions from their laptop as
part of the Installation Wizard process.
Installation of the BRI Media Module is supported.
The Installation Wizard supports installation of a G700 Media Gateway with a Local
Survivable Processor (LSP).
Remote G700s without an Internal Call Controller (ICC) Media Module can be configured
using the Installation Wizard by temporarily installing a spare ICC Media Module in the
G700 Media Gateway until the end of the installation process.
Provide an Electronic pre-Installation Worksheet (EIW) to automate the task of importing
selected pre-installation data. This capability is focused on importing IP address
information.
A customization template to allow for the selective customization of defaulted translation
data.
Support for Japan, United Kingdom, and France, including system and trunk level
parameters. May be extended to Australia and other countries prior to the next release of
Communication Manager.
Support configuration of IP trunks.
Issue 6 May 2009
27
Communication Manager Overview
l
l
Support trunk diagnostics.
Support IP address configuration of distributed G700 gateways through the Gateway
Installation Wizard (GIW).
Updates to the Avaya Installation Wizard are available on the Web, and are not necessarily
linked to any software release of Communication Manager. The latest version of Avaya
Installation Wizard can be downloaded from http://support.avaya.com/avayaiw.
Gateway Installation Wizard
The Gateway Installation Wizard (GIW) is a separate application that assists in installing and
configuring certain media gateways. For more information, see your Avaya representative.
Updates to GIW are available on the Web, and are not necessarily linked to any software
release of Communication Manager. The latest version of GIW can be downloaded from
http://support.avaya.com/avayaiw.
A common Electronic Pre-installation Worksheet to collect and import configuration data for all
supported media gateways, and support for Electronic Pre-installation Worksheet in Excel 2007
file format are available.
System Management Interface
You can perform the Avaya S8xxx Server tasks using the System Management Interface.
A top navigation menu is provided on all System Management Interface screens. It provides the
Help and Log Off options as well as the menu entries for all system management tasks
available for the current session/login. Access to other servers, part of the same system setup,
may also be available in the top navigation menu if the server is configured to support this. The
main menu items present sub-menu entries with a mouse rollover. Those entries provide
access to the various interfaces available to the session. Selecting a sub-menu item directs the
session to a system management application. More information about each sub-menu entry is
displayed with a mouse rollover of the entry.
28 Avaya AuraTM Communication Manager Overview
Avaya 8XXX Servers and media gateways
Avaya 8XXX Servers and media gateways
For information about any of the Avaya 8xxx Servers or media gateways that can run
Communication Manager, see Avaya Aura™ Communication Manager Hardware Description
and Reference, 555-245-207.
Also see the capacities table for the entire list of updated capacities. The most up-to-date
system capacity information is not listed in Communication Manager documentation. Instead,
this information is available online. See Capacities on page 26 for instructions how to locate the
capacities table.
Multi-Tech gateway support
Communication Manager supports a voice over IP (VoIP) gateway from Multi-Tech Systems,
Inc., a third-party vendor. Any system that is running Communication Manager can connect and
run this Multi-Tech gateway.
For more information, see Avaya Aura™ Communication Manager Hardware Description and
Reference, 555-245-207.
Co-residency of Communication Manager
and SIP Enablement Services
The Co-Residency of Communication Manager and Avaya Aura™ SIP Enablement Services is
a solution that helps reduce the cost of adding full, native support for SIP to your
communications network by merging the hardware platforms of the Communication Manager
software. The merged platform for co-residency is the Avaya S8300C Server, with compact
flash replacing RAMDISK.
The co-resident hardware platform for Avaya Aura™ SIP Enablement Services and
Communication Manager enables the two to operate more efficiently (for example, without
firewall issues or the need to encrypt links between the two), and to share some of the same
server resources and capabilities. Also, some of the pages of the System Management
Interface are shared and reused, including the web pages for system backup and restore
capabilities. The system logging, process status and role-based web access controls also are
the same for both.
You can install Avaya Aura™ Communication Manager Messaging voice mail with
Communication Manager on S8300 Servers. Both the Avaya Aura™ SIP Enablement Services
software and Communication Manager are installed, patched and configured in the usual way,
with a separate license for each. This makes transitioning to the new co-resident
implementation easier for existing administrators. Also like the Avaya Aura™ Communication
Manager Messaging, the Avaya Aura™ SIP Enablement Services software on S8300 Servers
Issue 6 May 2009
29
Communication Manager Overview
can be enabled or disabled and is disabled by default. A server reboot is required after enabling
SES on the S8300C.
The co-resident Avaya Aura™ SIP Enablement Services software is available with
Communication Manager deployed as a primary controller on the S8300C. The co-resident
Avaya Aura™ SIP Enablement Services is not currently available on any other hardware
platforms or if Communication Manager is deployed as an ESS or LSP.
The maximum number of SIP stations supported by Communication Manager on the S8300
depends on the configuration:
For Communication Manager primary controllers on the S8300A/B, the maximum number of
SIP stations is 450 or the capacity of the gateway being used with the S8300, if smaller.
For Communication Manager primary controllers on the S8300C with the co-resident Avaya
Aura™ SIP Enablement Services server disabled, the maximum number of SIP stations is 450
or the capacity of the gateway being used with the S8300, if smaller.
For Communication Manager primary controllers on the S8300C with the co-resident SES
server enabled, the maximum number of SIP stations is 400 or 200, depending on whether or
not the SIP stations are configured to use signaling encryption, or the capacity of the gateway
being used with the S8300C, if smaller.
When Avaya Aura™ SIP Enablement Services is deployed co-resident with Communication
Manager, the server must be configured as a Home server or combined Home/Edge server.
Co-resident Avaya Aura™ SIP Enablement Services cannot be configured as a standalone
Edge server, and therefore cannot perform the core routing function in an Avaya Distributed
Office solution. A standalone S8500-based Avaya Aura™ SIP Enablement Services Edge 5.0 is
still required for these larger SIP implementations.
For a complete description of this feature, see Administering Avaya Aura™ SIP Enablement
Services on the Avaya S8300 Server, 03-602508.
30 Avaya AuraTM Communication Manager Overview
Chapter 2: Application programming interface
An application programming interface (API) allows numerous software applications to work with
Communication Manager. APIs also allow a client programmer to create their own applications
that work with Communication Manager.
Application Enablement Services
Application Enablement Services (AE Services) is a connector that provides connectivity
between applications and Communication Manager. This connector allows development of new
applications and new features without having to modify Communication Manager or expose its
proprietary interfaces.
AE Services provides a single common platform architecture for call control, device control,
media control, and management. AE Services enables internal Avaya developers and external
partners to create powerful applications that harness the extensive Communication Manager
feature set.
Note:
AE Services has its own set of customer documentation, including an overview.
This Overview of Communication Manager does not outline the changes to AE
Services.
Note:
Avaya provides two different AE Services deployment options:
l
Software-only option
l
Bundled server option
The same client applications and software development kits (SDK) can run against both
options.
Software-only option
Avaya provides the AE Services software, which is the AE Services connector server software
and the AE Services SDK. The customer obtains the prerequisite hardware, platform software,
and third-party software. The customer then installs and maintains all software and hardware.
AE Services requires an AE Services license file. The license file can only be accessed by
Avaya Services or by an Avaya BusinessPartner.
Issue 6 May 2009
31
Application programming interface
Bundled server option
Avaya provides the hardware server and all of the necessary software:
l
AE Services connector server software
l
AE Services SDK
l
Platform
l
Third-party software
Avaya service technicians install the hardware and software. If the customer buys a
maintenance and/or a service contract, Avaya also provides maintenance and/or service for the
system. Otherwise, the customer maintains the system.
CVLAN
CallVisor LAN (CVLAN) is an application programming interface (API) that enables applications
to communicate with Communication Manager. CVLAN sends and receives ASAI messages
over shared ASAI links on TCP/IP. An application can use ASAI messages to monitor and
control Communication Manager resources.
CVLAN software consists of a client component and a server component. The CVLAN client
can be installed on a server or on a client workstation. The CVLAN client provides clients with
access to the switch using the CVLAN server.
Web services
Telephony Service
Telephony Service (TS) is a web service that exposes basic outbound call control features of
Communication Manager. Telephony Service enables its clients to originate an outbound call,
drop a call, transfer a call, or conference a party into a call.
Telephony Service is one of the web services that resides on the Application Enablement
Services platform (AE Services).
System Management Service
System Management Service (SMS) is a web service that exposes management features of
Communication Manager to clients. SMS enables its clients to display, list, add, change, and
remove specific managed objects on Communication Manager that are available through the
OSSI protocol and SAT screens.
32 Avaya AuraTM Communication Manager Overview
Device and media control API
SMS is one of the web services that resides on the Application Enablement Services platform
(AE Services).
User Service
User Service provides a common way of administering, retrieving, and programmatically
operating on user data. User Service provides a common user store and a programmable
interface for products and applications with which to integrate. User Service has a common
industry-standard data store (LDAP) as the repository for common user profile data.
User Service has web services as the infrastructure. This infrastructure allows products to
integrate with User Service at your schedule. User Service exposes a programmatic SOAP
interface that allows clients to write third party applications to utilize its functionality.
This integration occurs through the use of software adapters to User Service. The adapter and
web services technology allows User Service to publish user events to the product spaces, and
the product spaces to publish events to the common user area.
So if an administrator adds a user to the common store, a user event is sent to all participating
products with the appropriate information. Likewise, if a product level administrator modifies a
user record in its own user system, an event is sent to User Service for the modified data to be
stored in the common store. User Service then relays this user event to the other participating
product areas.
Device and media control API
Device and media control API provides a connector to Communication Manager that allows
clients to develop applications that provide first party call control. Applications can register as IP
extensions on Communication Manager and then monitor and control those extensions.
Device and media control API consists of connector server software and a connector client API
library. The connector server software runs on a hardware server that is independent from
Communication Manager. That is, device and media control API does not run co-resident with
Communication Manager.
Ask your Avaya representative for a complete list of device and media control API
documentation.
Issue 6 May 2009
33
Application programming interface
DEFINITY LAN Gateway
DEFINITY LAN Gateway (DLG) is a software service that tunnels the ASAI call control protocol
messages onto IP packets for transport between a customer Computer Telephony Integration
(CTI) server or application and Communication Manager.
Adjunct switch application interface
Adjunct Switch Application Interface (ASAI) links Communication Manager and adjunct
applications. The interface allows adjunct applications to access Communication Manager
features and supply routing information to the system.
ASAI is the Avaya recommendation for Computer Telephony Integration (CTI). ASAI is based
on the Q.932 protocol.
JTAPI
Java telephony application programming interface (JTAPI) is an open API supported by Avaya
computer telephony that enables integration to Communication Manager ASAI. It is an
object-oriented programming interfaces favored for the development of multimedia solutions.
JTAPI applications are supported on any clients that supports a JAVA virtual machine (this
includes Windows, UnixWare, and Solaris platforms), or a Java-compatible Web browser.
TSAPI
Telephony Services Application Programming Interface (TSAPI) is an open API supported by
Avaya computer telephony that allows integration to Communication Manager ASAI.
TSAPI is based on international standards for CTI telephony services. Specifically, the
European Computer Manufacturers Association (ECMA) CTI standard definition of
Computer-Supported Telecommunications Applications (CSTA) is the foundation for TSAPI.
The CSTA standard is a technical agreement reached by an open, multi-vendor consortium of
major switch and computer vendors. Since CSTA Services and protocol definitions are the basis
for TSAPI, TSAPI provides a generic, switch-independent API. CSTA services logically
integrate the two most common pieces of equipment on user desktops, the telephone and the
personal computer.
34 Avaya AuraTM Communication Manager Overview
TSAPI
Security administration for telephony services allows administrators to restrict user access to
TSAPI features in various ways. For example, an administrator might restrict a user to control
and monitoring of the telephone at their desktop. Similarly, an administrator can restrict a user
to call control and monitoring of the telephone at any desktop where they log in.
Expanded security permissions can increase user control in support of work group or
departmental telephony applications. Administrators can expand user permissions even further
to include any telephone or device that it is possible to control on a CTI link. An administrator
might assign an unrestricted security permission level to a server application that processes
calls before call delivery to user desktops in a call center environment. An administrator can
assign different users different permissions.
Issue 6 May 2009
35
Application programming interface
36 Avaya AuraTM Communication Manager Overview
Chapter 3: Attendant features
Communication Manager contains many exciting features that provide easy ways to
communicate through your telephone system attendant (operator). In addition, attendants can
connect to their console (switchboard) from other telephones in your system, thereby expanding
the attendant capabilities.
Accessing the attendant
Dial access to attendant
The dial access to attendant feature allows you to reach an attendant by dialing an access
code. The attendant can then extend the call to a trunk or to another telephone.
Individual attendant access
Individual attendant access allows you to call a specific attendant console. Each attendant
console can be assigned an individual extension number.
Recall
This feature allows users to recall the attendant when they are on a two-party call or on an
attendant conference call held on the console.
l
l
Single-line users press the recall button or flash the switchhook to recall the attendant.
Multi-appearance users press the conference or transfer button to recall the attendant and
remain on the connection when either button is used.
Issue 6 May 2009
37
Attendant features
Attendant backup
The attendant backup feature allows you to access most attendant console features from one or
more specially-administered backup telephones. This allows you to answer calls more promptly,
thus providing better service to your guests and prospective clients.
When the attendant console is busy, you can answer overflow calls from the backup telephones
by pressing a button or dialing a feature access code. You can then process the calls as if you
are at the attendant console. The recommended backup telephones are the Avaya models
6408, 6416, or 6424.
Attendant room status
Communication Manager allows an attendant to see whether a room is vacant or occupied, and
what the housekeeping status of each room is. This feature is available only when you have
enhanced hospitality enabled for your system (see Hospitality on page 89).
This feature combines the property management capabilities of housekeeping status and
check-in/check-out, but does not require that you have a property management system (PMS).
Attendant functions using Distributed Communications
System protocol
Control of trunk group access
Control of trunk group access allows an attendant at any node in the Distributed
Communications System (DCS) to take control of any outgoing trunk group at an adjacent
node. This is helpful when an attendant wants to prevent telephone users from calling out on a
specific trunk group for any number of reasons, such as reserving a trunk group for incoming
calls or for a very important outgoing call.
38 Avaya AuraTM Communication Manager Overview
Call handling
Direct trunk group selection
Direct trunk group selection allows the attendant direct access to an idle outgoing trunk in a
local or remote trunk group by pressing the button assigned to that trunk group. This feature
eliminates the need for the attendant to memorize, or look up, and dial the trunk access codes
associated with frequently used trunk groups. Direct trunk group selection is intended to
expedite the handling of an outgoing call by the attendant.
Inter-PBX attendant calls
Inter-PBX attendant calls allows attendants for multiple branches to be concentrated at a main
location. Incoming trunk calls to the branch, as well as attendant-seeking voice-terminal calls,
route over tie trunks to the main location.
Call handling
Attendant Intrusion
Use the Attendant Intrusion feature to allow an attendant to intrude on an existing call. The
Attendant Intrusion feature is also called Call Offer.
Attendant lockout - privacy
This feature prevents an attendant from re-entering a multiple-party connection held on the
console unless recalled by a telephone user. This feature is administered on a system-wide
basis. It is either activated or not activated.
Attendant split swap
The attendant split swap feature allows the attendant to alternate between active and split calls.
This operation may be useful if the attendant needs to transfer a call but first must talk
independently with each party before completing the transfer.
Issue 6 May 2009
39
Attendant features
Attendant vectoring
Attendant vectoring provides a highly flexible approach for managing incoming calls to an
attendant. For example, with current night service operation, calls redirected from the attendant
console to a night station can ring only at that station and will not follow any coverage path.
With attendant vectoring, night service calls will follow the coverage path of the night station.
The coverage path could go to another station and eventually to a voice mail system. The caller
can then leave a message that can be retrieved and acted upon.
Automated attendant
Automated attendant allows the calling party to enter the number of any extension on the
system. The call is then routed to the extension. This allows you to reduce cost by reducing the
need for live attendants.
Backup alerting
The backup alerting feature notifies backup attendants that the primary attendant cannot pick
up a call. It provides both audible and visual alerting to backup stations when the attendant
queue reaches its queue warning level. When the queue drops below the queue warning level,
alerting stops.
Audible alerting also occurs when the attendant console is in night mode, regardless of the
attendant queue size.
Call waiting
Call waiting allows an attendant to let a single-line telephone user who is on the telephone know
that a call is waiting. The attendant is then free to answer other calls. The attendant hears a call
waiting ringback tone and the busy telephone user hears a call waiting tone. This tone is heard
only by the called telephone user.
Calling of inward restricted stations
A telephone with a class of restriction (COR) that is inward restricted cannot receive public
network, attendant-originated, or attendant-extended calls. This feature allows you to override
this restriction.
40 Avaya AuraTM Communication Manager Overview
Call handling
Conference
The conference feature allows an attendant to set up a conference call for as many as six
conferees, including the attendant. Conferences from inside and outside the system can be
added to the conference call.
Starting with Communication Manager release 3.0, attendants can set up conferences for more
than six people using the Enhanced Meet-me Conferencing feature. For more information, see
Expanded Meet-me Conferencing on page 75.
Enhanced Return Call to (same) Attendant
Communication Manager provides individual queuing functions for each attendant supporting a
multiplicity of waiting calls at a given time. When at least one call is waiting in the above queue,
the Individual Calls Waiting Indicator is displayed in red. Calls queue as long as the attendant is
busy.
When the feature Enhanced Return Call to (same) Attendant is enabled, a call returning to a
busy or now unavailable same attendant is placed into an individual waiting queue for this
attendant, instead of queuing it into the attendant’s group. A waiting Return Call moves from the
attendant’s queue to the queue for the attendant group after a certain period of time.
Listed directory number
Allows outside callers to access your attendant group in two ways, depending on the type of
trunk used for the incoming call. You can allow attendant group access through incoming direct
inward dial trunks, or you can allow attendant group access through incoming central office and
foreign exchange trunks.
Override of diversion features
The override of diversion feature allows an attendant to bypass diversion features such as send
all calls and call coverage by putting a call through to an extension even when these diversion
features are on. This feature, together with attendant intrusion, can be used to get an
emergency or urgent call through to a telephone user.
Issue 6 May 2009
41
Attendant features
Priority queue
Priority queue places incoming calls to the attendant in an orderly queue when these calls
cannot go immediately to the attendant. This feature allows you to define twelve different
categories of incoming attendant calls, including emergency calls, which are given the highest
priority.
Release loop operation
Release loop operation allows the attendant to hold a call at the console if the call cannot
immediately go through to the person being called. A timed reminder begins once the call is on
hold. If the call is not answered within the allotted time, the call returns to the queue for the
attendant. Timed reminders attempt to return the call to the attendant who previously handled it.
Only when the original attendant is unavailable are calls returned to the queue.
Selective conference mute
See Selective conference mute on page 77.
Serial calling
The serial calling feature enables an attendant to transfer trunk calls that return to the same
attendant after the called party hangs up. The returned call can then transfer to another station
within the switch. This feature is useful if trunks are scarce and direct inward dialing services
are unavailable. An outside caller may have to redial often to get through because trunks are so
busy. Once callers get through to an attendant they can use the same line into the switch for
multiple calls. The attendant display shows if an incoming call is a serial call.
Timed reminder and attendant timers
Attendant timers automatically alert the attendant after an administered time interval for the
following types of calls:
l
Extended calls to be answered or waiting to be connected to a busy single-line telephone
l
One-party calls placed on hold on the console
l
Transferred calls that have not been answered after transfer
42 Avaya AuraTM Communication Manager Overview
Centralized Attendant Service
The timed reminder feature informs the attendant that a call requires additional attention. After
the attendant reconnects to the call, the user can either choose to try another extension
number, hang up, or continue to wait. Communication Manager supports a variety of
administrable attendant timers for use in a variety of situations.
Centralized Attendant Service
Centralized Attendant Service (CAS) enables attendant services in a private network to be
concentrated at a central location. Each branch in a centralized attendant service has its own
listed directory number or other type of access from the public network. Incoming calls to the
branch, as well as calls made by users directly to the attendants, are routed to the centralized
attendants over release link trunks.
Display
The display feature shows call-related information that helps the attendant to operate the
console. This feature also shows personal service and message information. Information is
shown on the alphanumeric display on the attendant console. Attendants may select one of
several available display message languages: English, French, Italian, or Spanish. In addition,
your company may define one additional language for use by users and attendants on their
display.
Increased attendant consoles
Depending on the platform, or server, you are using, Communication Manager supports an
increase of attendant consoles. For example, an S8720XL platform supports 414 attendants,
while a regular S8720 platform supports 128 attendants. This limit applies to both attendant
console telephones with direct extension selectors (DXS), as well as soft attendant consoles.
The attendant consoles may collectively manage a single defined call load. Alternately, each
attendant console may be defined so that it manages its own unique call load.
Issue 6 May 2009
43
Attendant features
Making calls
Auto Start and Do Not Split
The Auto Start feature allows the attendant to make a telephone call without pushing the start
button first. If the attendant is on an active call and presses digits on the keypad, the system
automatically splits the call and begins dialing the second call.
The Do Not Split feature deactivates the auto start feature and allows the sending of touch
tones over the line for the purposes of such things as picking up messages.
Auto Manual Splitting
Auto Manual Splitting allows an attendant to announce a call or consult privately with the called
party without being heard by the calling party on the call. It splits the calling party away so the
attendant can confidentially determine if the called party can accept the call.
Monitoring calls
Attendant control of trunk group access
Use the Attendant Control of Trunk Group Access feature to allow the attendant to control
outgoing and two-way trunk groups. The attendant usually activates this feature during periods
of high use. This is helpful when an attendant wants to prevent telephone users from calling out
on a specific trunk group. Some reasons are to reserve a trunk group for incoming calls or for a
very important outgoing call.
This feature also prevents telephone users from directly accessing an outgoing trunk group that
the attendant has controlled.
44 Avaya AuraTM Communication Manager Overview
Monitoring calls
Attendant direct extension selection
This feature allows the attendant to keep track of extension status - whether the extension is
idle or busy - and to place or extend calls to extension numbers without having to dial the
extension number. The attendant can use this feature in two ways:
l
using standard direct extension selection access
l
using enhanced direct extension selection access
Attendant direct trunk group selection
With this feature, the attendant directs access to an idle outgoing trunk by pressing the button
assigned to the trunk group. This feature eliminates the need for the attendant to memorize, or
look up, and dial the trunk access codes associated with frequently used trunk groups. Pressing
a labelled button selects an idle trunk in the desired group.
Crisis alerts to an attendant console
Crisis alert uses both audible and visual alerting to notify attendant consoles when an
emergency call is made. Audible alerting sounds like an ambulance siren. Visual alerting
flashes the CRSS-ALRT button lamp and the display of the caller name and extension (or
room). The display of the origin of the emergency call enables the attendant or other user to
direct emergency service response to the caller. Though often used in the hospitality industry, it
can be set up to work with any standard attendant console.
When crisis alerting is active, the console is placed in position-busy mode so that other
incoming calls can not interfere with the emergency call notification. The console can still
originate calls to allow notification of other personnel. Once a crisis alert call has arrived at a
console, the console user must press the position-busy button to unbusy the console, and press
the crisis-alert button to deactivate audible and visual alerting.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the order
in which they were made. If the system is administered so that only one user must respond, the
first crisis alert remains active at the telephone where it was acknowledged. Subsequent calls
are queued to the next available station in the order in which they were made.
Issue 6 May 2009
45
Attendant features
Trunk group busy/warning indicators to attendant
This feature provides the attendant with a visual indication that the number of busy trunks in a
group has reached an administered level. A visual indication is also provided when all trunks in
a group are busy. This feature is particularly helpful to show the attendant that the attendant
control of trunk group access feature needs to be invoked.
Trunk identification by attendant
Trunk identification allows an attendant or display-equipped telephone user to identify a specific
trunk being used on a call. This capability is provided by assigning a trunk ID button to the
attendant console or telephone. This feature is particularly helpful for identifying a faulty trunk.
That trunk can then be removed from service and the problem quickly corrected.
Visually Impaired Attendant Service
Visually Impaired Attendant Service (VIAS) provides voice feedback to a visually impaired
attendant. Each voice phrase is a sequence of one or more single-voiced messages. This
feature defines six attendant buttons to aid visually impaired attendants:
l
l
l
Visually impaired service activation/deactivation button: activates or deactivates the
feature. All ringers previously disabled (for example, recall and incoming calls) become
reenabled.
Console status button: voices whether the console is in position available or position busy
state, whether the console is a night console, what the status of the attendant queue is,
and what the status of system alarms is.
Display status button: voices what is shown on the console display. VIAS support is not
available for all display features (for example, class of restriction information, personal
names, and some call purposes).
l
Last operation button: voices the last operation performed.
l
Last voiced message button: repeats the last voiced message.
l
Direct trunk group selection status button: voices the status of an attendant-monitored
trunk group.
The visually impaired attendant may use the Inspect mode to locate each button and determine
the feature assigned to each without actually executing the feature.
46 Avaya AuraTM Communication Manager Overview
Chapter 4: Call Center
The Avaya Aura™ Call Center provides a fully integrated telecommunications platform that
supports a powerful assortment of features, capabilities, and applications designed to meet all
of your customers' call center needs.
For a more complete description of Call Center features for various releases of Communication
Manager, see the following documents:
l
What's New in Avaya Aura™ Call Center 5.2
l
Avaya Aura™ Call Center 5.2 Automatic Call Distribution Reference
l
Avaya Aura™ Call Center 5.2 Call Vectoring and Expert Agent Selection (EAS) Reference
l
Avaya Business Advocate
Computer Telephony Integration
Computer Telephony Integration (CTI) enables Communication Manager features to be
controlled by external applications, and allows integration of customer databases of information
with call control features.
Avaya Computer Telephony (formally named CentreVu™ Computer Telephony) is server
software that integrates the premium call control features of Communication Manager with
customer information in customer's databases. It is a local area network (LAN)-based CTI
solution consisting of server software that runs in a client/server configuration. Avaya Computer
Telephony delivers the CTI architecture and platform that supports contact center application
requirements, along with emerging applications programming interfaces (APIs).
Adjunct route support for network call redirection
This feature provides the capability to invoke Network Call Redirection (NCR) through the route
request response to an adjunct route vector step. This allows a CTI application to directly utilize
NCR for redirecting an incoming call in the PSTN through the ASAI adjunct routing application.
The redirection request, along with the PSTN redirected to a telephone number, is included in
the route select message from the adjunct. The redirect request invokes whatever form of
network redirection that is assigned to the trunk group for the incoming call in the same manner
as a vector invoked NCR. Information forwarding to the redirected destination is supported in
the same manner as a vector invoked NCR.
Issue 6 May 2009
47
Call Center
This capability functions with either the network transfer type where the switch sets up the 2nd
leg of a call, or the network deflection type where the PSTN sets up the 2nd leg of a call of NCR
protocols.
Co-resident DEFINITY LAN Gateway
In simplest terms, the DEFINITY Local Area Network (LAN) Gateway, or DLG, is an application
that enables communications between TCP/IP clients and Communication Manager call
processing. In more technical terms, the DLG application is software that both routes
internetwork messages from one protocol to another (ISDN to TCP/IP) and bridges all ASAI
message traffic by way of a TCP/IP tunnel protocol.
In previous configurations, a DEFINITY LAN gateway (DLG) was connected externally on a
separate TN801 MAPD circuit pack. The DLG application is packaged internally where it
co-resides with the Communication Manager. The internally packaged DLG is referred to as the
co-resident DLG.
Co-resident DLG is only available with the S8300 Server.
Co-resident DLG provides the functionality of the Adjunct/Switch Application Interface (ASAI)
using an ethernet transport instead of a Basic Rate Interface (BRI) transport. In the S8300
Server, connectivity is provided through the processor ethernet.
For more information on co-resident DLG and the G700 Media Gateway, see chapters
“DEFINITY LAN Gateway and ASAI-Ethernet,” and “Installation and Test for CallVisor ASAI,” in
the Avaya MultiVantage Software CallVisor ASAI Technical Reference.
Also see the following documents:
l
DEFINITY Enterprise Communications Server CallVisor ASAI Applications Over MAPD
l
Installation for Adjuncts and Peripherals for Avaya AuraTM Communication Manager
Direct Agent Announcement
Direct Agent Announcement (DAA) enhances direct agent calling capabilities for Adjunct Switch
Application Interface (ASAI) and Expert Agent Selection (EAS). It plays an announcement to
direct agent callers waiting in a queue.
Flexible billing
The flexible billing feature allows Communication Manager or an adjunct to communicate with
the public network using ISDN PRI messages to change the billing rate for an incoming
900-type call. Rate-change requests to specify a new billing rate can be made anytime after a
call is answered and before it disconnects.
48 Avaya AuraTM Communication Manager Overview
Computer Telephony Integration
Flexible billing is available in the U.S. for use with AT&T MultiQuest 900 Vari-A-Bill service.
Flexible billing requires an adjunct switch application interface and other application software.
Pending work mode change
This feature allows ASAI applications to change the current work mode of an agent while that
agent is busy on a call. The change is a pending change that will take effect as soon as all the
current calls are cleared.
Trunk group identification
Trunk group identification provides ASAI applications with the capability to obtain trunk group
information even when the Calling Party Number (CPN) is known. ASAI will provide the trunk
group information in the event reports for both inbound and outbound calls. If the Automatic
Number Identification (ANI) is known, the event reports will contain the trunk group information
and the CPN.
User-to-User Information propagation during
manual transfer/conference operations
This feature enables UUI, specifically used by ASAI, to be propagated to the new call during a
manual transfer or conference operation. Previously, ASAI UUI could not be sent in a setup
message when the call was transferred to another system, so the ASAI UUI was never passed
to an application monitoring calls on the system receiving the transfer.
This feature only applies to manual transfer and conference operations. If the transfer or
conference operation is controlled by a software application (for example, controlling calls or
agents over an ASAI link), the application can insert the desired ASAI UUI into the new call.
Block CMS Move Agent events
This feature lets you prevent the system from sending the ASAI logout-login event messages,
that are related to an agent move. When this CTI link option is activated, the changes to the
agent state, such as logout followed by login and return to previous state, will not be reported to
the ASAI adjunct. This operation is required by Avaya IC since the initial logout causes IC to
permanently logout the agent, disrupting normal operation. IC does not need to be informed of
agent skill moves via this method. This option will be available to other applications for use
where needed.
Issue 6 May 2009
49
Call Center
VDN override for ASAI messages
This feature provides a VDN option to override the called number in certain ASAI messages for
ISDN calls. This applies to CTI applications that require the active VDN extension instead of the
called number. This is a field on page 2 of the VDN Screen - “VDN Override for ISDN Trunk
ASAI Messages. The default value is no.
For calls to VDNs with the option set to y(es), the called number provided will correspond to the
active VDN for call instead of the original called number provided in the incoming ISDN SETUP
message. This applies to the ASAI call-offered, alerting, queued and connect event messages
and the adjunct route-request message.
Automatic Call Distribution
Automatic Call Distribution (ACD) is the basic building block for call center applications. ACD
offers you a method for distributing incoming calls efficiently and equitably among available
agents. With ACD, incoming calls can be directed to the first idle or most idle agent within a
group of agents.
Agents in an ACD environment are assigned to a hunt group, a group of agents handling the
same types of calls. A hunt group is also known as a split or skill with Expert Agent Selection
(EAS).
A hunt group is especially useful when you expect a high number of calls to a particular
telephone extension. A hunt group might consist of people trained to handle calls on specific
topics. For example, the group might be:
l
A benefits department within your company
l
A service department for products you sell
l
A travel reservations service
l
A pool of attendants
In addition, a hunt group might consist of a group of shared telecommunications facilities. For
example, the group might be:
l
A modem pool
l
A group of data-line circuit ports
l
A group of data modules
In the following example (Figure 2: A basic example of automatic call distribution on page 51),
hunt group “A” receives calls only when agents are available since it has no queue. Calls to hunt
group “B” can be queued while agents are unavailable, and redirected to hunt group “C” if not
50 Avaya AuraTM Communication Manager Overview
Automatic Call Distribution
answered within an administrable time. Calls to hunt group “C” are redirected to voice mail if not
answered within an administrable time.
Figure 2: A basic example of automatic call distribution
2
1
3
4
5
7
6
6
8
cydfauto KLC 030102
Figure notes:
1.
System running Avaya AuraTM
Communication Manager
5.
Hunt group C: general information
2.
Incoming lines
6.
Queues
3.
Hunt group A: business travel
7.
Call coverage to hunt group C
4.
Hunt group B: personal travel
8.
Voice mail
Abandoned Call Search
Abandoned Call Search allows a central office that does not provide timely disconnect
supervision to identify abandoned calls. An abandoned call is one in which the calling party
hangs up before the call is answered. Abandoned Call Search is suitable only for older central
offices that do not provide timely disconnect supervision.
Issue 6 May 2009
51
Call Center
Interruptible Aux work
If a skill’s designated service level is not met, this feature can make available the unavailable
EAS agents who are in Auxiliary (AUX) work mode and have an interruptible reason code.
Using this feature, for example, during the call volume spikes, you can use agents in Auxiliary
(AUX) work mode to achieve the desired service level.
Adjunct Routing
Adjunct Routing is a vector step that, when executed, sends a route request over the specified
link to the connected adjunct asking where to route the call being processed. The adjunct is
then to respond with a route-select message specifying the destination either internal or outside
number where the call is to be routed. Adjunct Routing is used in conjunction with ASAI.
Auto-Available Split
Auto-Available Split (AAS) allows members of an Automatic Call Distribution (ACD) split to be
continuously in auto-in work mode. An agent in auto-in work mode becomes available for
another ACD call immediately after disconnecting from an ACD call. You can use AAS to bring
ACD-split members back into auto-in work mode after a system restart.
Although not restricted to such, this feature is intended to be used for splits containing only
recorders or voice-response units.
Automatic Number Identification
Use the Automatic Number Identification (ANI) feature to display telephone number of the
calling party on your display telephone. The system uses ANI to interpret calling party
information that is signaled over multifrequency (MF) or Session Initiation Protocol (SIP) trunks.
Incoming Automatic Number Identification
Use inband signaling for information, such as the address digits for the called party, that is
delivered over the same trunk circuit that is used for the voice or data connection. Use
out-of-band or ISDN signaling when signaling information passes through a different signaling
path than the path that is used for the voice or data connection.
For example, when a call is made from 555-3800 to your display telephone at extension 81120,
and the Incoming Tone (DTMF) ANI field is set to *ANI*DNIS* on the Trunk Group screen,
52 Avaya AuraTM Communication Manager Overview
Automatic Call Distribution
your trunk group receives *5553800*81120*. If the same field is set to ANI*DNIS*, your trunk
group receives 5553800*81120*. In both cases, call from 555-3800 appears on your telephone
display.
If you do not use inband ANI, the incoming trunk group name appears on your telephone
display.
Outgoing Automatic Number Identification
Outgoing ANI applies to outgoing Russian MF ANI, R2-MFC ANI, China #1 MF ANI, and Spain
Multi Frequency España (MFE) ANI trunks only.
Use Outgoing ANI to specify the type of ANI to send on outgoing calls. You can define MF ANI
(the calling party number, sent through multifrequency signaling trunks) prefixes by COR. This
allows a system to send different ANIs to different central offices (COs).
For a tandem call that uses different types of incoming and outgoing trunks, the server uses:
l
The COR-assigned call type of the incoming trunk for Russian or R2-MFC outgoing trunks
l
Automatic Route Selection (ARS) call types for MFE outgoing trunks
Local feedback for queued ACD calls
A cost saving trend used by many call centers is the movement of agent seats from locations in
the US and EU to offshore locations. One detriment to achieving these savings is the increase
in trunk costs by redirecting calls to these offshore locations.
When a call is rerouted to an alternate switch, it becomes the responsibility of the destination
switch to provide audible feedback to the caller while that call remains in queue at the
destination switch waiting for an available agent. Typically, such audible feedback takes the
form of music interspersed with recorded announcements.
When the trunks between the sending and receiving switches are IP trunks, bandwidth is
utilized when the music and recorded announcement packets are sent from the destination
switch to the caller. Because of the continuous nature of music, the bandwidth required to
provide this audible feedback to callers in queue is generally greater than that required to
support a conversation between a caller and an agent.
Communication Manager allows vector processing to continue at the local sending switch, even
after a call has been routed to a queue on an offshore destination switch. Vector processing at
the sending switch can then continue to provide audible feedback to the caller while the call is in
queue at the destination switch. No packets need be sent over the IP trunk during the queuing
phase of the call.
Issue 6 May 2009
53
Call Center
Queue status indicators
Communication Manager allows you to assign queue status indicators for ACD calls based on
the number of calls in queue and the time in queue. To help monitor queue activity, you can
assign these indications to lamps on agent, supervisor, or attendant terminals, or on consoles.
In addition, you can define auxiliary queue warning lamps to track queue status. On display
telephones, you can display the number of calls in queue, and the time in queue of the oldest
call.
Avaya Basic Call Management System
The Avaya Basic Call Management System (BCMS) helps you fine tune your call center
operation by providing reports with the data necessary to measure your call center agents
performance.
The BCMS feature offers call management control and reporting at a low cost for call centers of
up to 2000 agents. BCMS collects and processes ACD call data (up to seven days) within the
system; an adjunct processor is not required to produce call management reports.
The following are the types of reports that can be generated:
l
Real-time reports, such as:
- Agent status
- System status
- Vector directory number status
l
Historical reports, such as:
- Agent
- Agent summary
- Split
- Split summary
- Trunk group
- Vector directory number
54 Avaya AuraTM Communication Manager Overview
Avaya Basic Call Management System
Avaya Business Advocate
Avaya Business Advocate is the collection of features that provide flexibility in the way a call is
selected for an agent in a call surplus situation, and in the way an agent is selected for a call.
Instead of the traditional “first in, first out” approach, the needs of the caller, potential business
value, and the desire to wait are calculated. The system then decides what agents should be
matched to the callers.
Auto reserve agents
Auto reserve agents allows the system to use the percent allocation distribution feature for
agent skills.
Call selection override per skill
Call selection override is determined by skill. Call center supervisors can override the normal
call handling activity either on particular skills only, or for the entire call center.
Dynamic percentage adjustment
The dynamic percentage adjustment feature allows the system to compare actual levels of
service with service targets. The system can then adjust the service target so that the overall
use of the skill is more efficient.
Dynamic queue position
Dynamic queue position allows the system to put calls from multiple vector directory numbers
(VDNs) into a skill queue. The calculation is based on the ratio of ASA for the VDNs being equal
to the ratio of service objectives for the VDNs. This feature ensures balanced call handling
across VDNs.
Dynamic threshold adjustment
Dynamic threshold adjustment allows the system to compare actual levels of service with
service targets, and to adjust overload thresholds. This feature makes the use of overload
agents more efficient.
Logged-in advocate agent counting
The logged-in advocate agent counting feature counts agents toward the advocate agent limit if
a service objective, percent allocation, or a reserved skill is assigned to the agent login ID, or if
one of the agent skills is assigned least occupied agent or service level supervisor.
Issue 6 May 2009
55
Call Center
Percent allocation distribution
Percent allocation distribution allows the system to distribute calls to auto reserve agents by
comparing a reserve agent work time in a skill with the target allocation for that skill.
Reserve agent time in queue activation
This feature activates a reserve agent either if the expected wait time (EWT) exceeds a
pre-determined threshold, or if the call time in the queue exceeds the administered service level
supervisor threshold. Reserve agents are then dropped off a skill only when both of the
following conditions are met:
l
The EWT for the skill drops below both administered thresholds.
l
The head call time in queue no longer exceeds the service level supervisor threshold.
Avaya Call Center features supported on the Avaya G700
Media Gateway
Avaya Call Center functionality is supported on the G700 Media Gateway with Communication
Manager, with either an S8300 Server or an S87XX Server.
The Avaya S8300 Server or S87XX Server with the Avaya G700 Media Gateway provides
Avaya Call Center “Basic” software (included with Communication Manager) capability and
optional Computer Telephony Integration (CTI) as a lower-cost call center solution for small or
branch offices. For the latest capacities of supported number of agents and media gateways,
please see the capacities document available at http://www.avaya.com/support. See
Capacities on page 26 for instructions how to locate the capacities document.
The Avaya G700 Media Gateway with the Avaya S8300 Server supports more robust call center
capabilities including Avaya Call Center “Deluxe,” which supports Avaya Best Service Routing
and optional Avaya Virtual Routing, and Avaya Call Center “Elite,” which features Avaya Expert
Agent Selection and services as the foundational software for the optional Avaya Business
Advocate and Avaya Dynamic Advocate software.
The call center capabilities found in either optional software package (Deluxe or Elite) allow
Communication Manager Call Center customers to enhance their customer service, help desk,
travel, and other operations by providing powerful, integrated call routing via “call vectoring” and
resources selection.
56 Avaya AuraTM Communication Manager Overview
Avaya Call Management System
Avaya Call Management System
The Avaya Call Management System (CMS) collects call traffic data, formats management
reports, and provides an administration interface for Automatic Call Distribution (ACD). It helps
you manage the people, traffic load, and equipment in an ACD environment by answering such
questions as:
l
How many calls are we handling?
l
How many callers abandon their calls before talking with an agent?
l
Are all agents handling a fair share of the calling load?
l
Are our lines busy often enough to warrant adding additional ones?
l
How has traffic changed in a given ACD hunt group over the past year?
Avaya virtual routing
Avaya virtual routing (formerly known as Look-Ahead Interflow or LAI) balances the load of ACD
calls across multiple locations. Virtual routing helps customers balance call loads among their
locations by analyzing demand and directing each call to the location best able to handle it - for
example, based on call volume, waiting time in queue, or the time of day.
With Avaya virtual routing, you can optionally route a call to a backup location based on your
system ability to handle the call within parameters defined in a vector. In turn, the backup
system can accept or deny the call also based on defined parameters.
Avaya virtual routing allows interflowing of only the call(s) at or near the head of the queue to
provide First In/First Out (FIFO) call distribution and significantly reduce call and trunk
processing for Avaya virtual routing.
Enhanced information forwarding
Enhanced information forwarding allows call center related information to be passed
transparently over some public networks and non-QSIG or QSIG private networks using
codeset 0 shared user-to-user information (UUI) (for non-QSIG) or QSIG manufacturer-specific
information (MSI). For more information on UUI, see User-to-user information over the public
network on page 70.
Issue 6 May 2009
57
Call Center
Call center release control
Call center release control determines which features are “active” on your switch. The call
center release control feature controls whether certain call center software features are
available to you.
Call prompting
Call prompting allows the system to collect information from the calling party and direct the calls
using call vectoring.
The caller is verbally prompted by the system and enters information in response to the
prompts. This information is then used to redirect the call or handle the call in some other way
(taking a message, for example). This feature is mostly used to enhance the efficient handling
of calls in the automatic call distribution application.
Data collection
Data collection allows the calling party to enter data that can then be used by a host computer
application to assist in call handling. For example, this data may be the calling party account
number, which could then be used to support an inquiry/response application.
Data In/Voice Answer
Data In/Voice Answer (DIVA) allows the calling party to hear selected announcements based on
the digits that he or she enters. This may be used for applications such as an audio bulletin
board.
58 Avaya AuraTM Communication Manager Overview
Call vectoring
Call vectoring
Call vectoring is a versatile method of routing incoming calls that can be combined with
automatic call distribution for maximum benefit and call center efficiency. A call vector is a
series of call processing steps (such as providing ringing tones, busy tones, music,
announcements, and queuing the call to an ACD hunt group) that define how calls are handled
and routed. The steps, called vector commands, determine the type of processing that specific
calls will receive.
Vector commands may direct calls to on-premises or off-premises destinations, to any skill or
hunt group, or to a specific call treatment such as an announcement, forced disconnect, forced
busy, or music.
With combinations of different vector commands, incoming callers can be treated differently
depending on the time or day of the call, the expected wait time (EWT), the importance of the
call, or other criteria. Each vector can have up to 32 commands. Communication Manager also
allows vectors to be linked through the “goto vector” command.
Advanced vector routing
Advanced vector routing is a collection of features that enhance Communication Manager
vector routing capabilities.
Average Speed of Answer routing
Average Speed of Answer (ASA) routing is an enhancement to call vectoring that provides a
flexible method for routing calls or queuing calls based on their average speed of answer for a
VDN or a split/skill.
Best service routing
Best service routing (BSR) distributes the call to the best local or remote split/skill among the
resources to be considered, based on expected wait time (EWT) or available agent
characteristics.
Best service routing polling over IP without B-channel
Best service routing (BSR) polling over IP without B-channel provides the ability to do BSR
polling between multiple sites over H.323 IP trunks without requiring an ISDN PRI B-channel.
This also eliminates the associated IP media processor hardware.
Issue 6 May 2009
59
Call Center
QSIG temporary signaling connections are used by the BSR polling software to eliminate the
need for the IP media processor board, thereby making BSR an even more cost effective
multi-site solution.
Expected Wait Time routing
The Expected Wait Time (EWT) feature makes call center routing decisions based on waiting
time for calls in queue, using a patented algorithm that continuously estimates call waiting
times. Announcements of the wait time customers can expect before their call is answered can
make time in queue more tolerable.
Call center messaging
Call center messaging gives the calling party the option of leaving a message or waiting in
queue for an agent. This may be used for an online order entry system or to further automate an
incoming call center operation.
Percentage allocation routing
This feature allows you to distribute calls among a set of call centers or VDNs based on
specified percent allocation. Various types of incoming calls that arrive at a particular VDN can
be directed to a Policy Routing Table (PRT) instead of to a vector. The PRT then distributes the
calls to the administered Route-to VDNs based on the specified percent allocation targets.
This feature is useful for segmented call-handling, outsourcing, and optimizing call handling in a
multiple-location enterprise.
Holiday vectoring
With holiday vectoring, a flexible approach for managing incoming calls on special dates is
available. Holiday vectoring allows for branching and routing of calls based on information about
special schedules. The special schedules are recorded in tables, each of which can hold up to
15 special dates or ranges of dates.
Vector Directory Number
Calls access Communication Manager vectors using Vector Directory Numbers (VDN). A VDN
is a “soft” extension number that is not assigned to a physical equipment location. A VDN has
several properties that are administered by the system manager.
60 Avaya AuraTM Communication Manager Overview
Call vectoring
A VDN can be accessed in almost any way that an extension can be accessed. When
answering a call, the answering agent sees the information (such as the name) associated with
the VDN on their display, and can respond to the call with knowledge of the dialed number. This
operation provides dialed number identification service (DNIS), allowing the agent to identify the
purpose of the incoming call.
Class of Restriction for VDN
Class of Restriction (COR) is checked for transfer to the VDN. It can also be used to block the
AUX trunk announcement from some agents. Observing can also be set to allow or restrict to
that VDN.
Display VDN for route-to DAC
Display VDN for route-to DAC provides a VDN option to have the display to the answering
agent show the “caller to VDN” format. The option for the “caller to VDN” display is required for
ACD applications where a call needs to be routed to a specific agent, and have the call go to
coverage if the agent doesn't answer or is logged out.
VDN in a coverage path
VDN in a coverage path enhances call coverage and call vectoring to allow you to assign vector
directory numbers as the last point in coverage paths. Calls that go to coverage can be
processed by vectoring/prompting to extend call coverage treatments.
VDN of origin announcement
VDN of origin announcement provides agents with a short message about the city of origin or
requested service or the caller, based on the VDN used to process the call. VOA messages
help agents respond appropriately to callers.
For example, if you have two 800 numbers, one for placing orders and one for technical
support, you can administer two VDNs to route calls to the same set of agents. When an
incoming call is routed to a VDN with a VOA assigned (for example, “new order” or “tech help”),
the VDN routes the call to a vector that can place the call in an agent queue. When an agent
answers the call, he or she hears the VOA message and can respond appropriately to the caller
request.
This feature is particularly useful for visually impaired agents or agents that do not have display
telephones.
VDN return destination
VDN return destination is an optional feature that re-routes a call that has been processed
through a vector, to the administered return destination. This step occurs once all parties,
except the originator, have dropped. The return destination must be a VDN extension.
Issue 6 May 2009
61
Call Center
Call Work Codes
Call Work Codes (CWC) allows ACD agents to enter digits for an ACD call to record the
occurrence of a customer-defined event, such as a social security numbers or telephone
numbers. The agent enters the call work code by operating the CWC feature button and using
the dial pad during an ACD (inbound) call without interrupting the conversation, or in the After
Call Work (ACW) mode following the call. The digits are displayed on a display-equipped
telephone while being entered.
Caller Information Forwarding
The Avaya call center also supports AT&T Caller Information Forwarding (CINFO) service,
allowing customers to collect customer-provided data forwarded through the network. This
information can be used to route calls or provide visual displays on agent voice terminals, or be
passed along to Computer Telephony Integration (CTI) applications.
Circular station hunt group
This hunt group type is an alternative to the “ddc” or “hot-seat” algorithm in a hunt group.
Communication Manager keeps track of the last extension in the hunt group that received a call.
When another incoming call arrives, it is sent to the next idle extension, bypassing the extension
that had received the previous call.
The first extension in the hunt group will no longer be the busiest telephone while the others in
the group are sitting idle.
Clear the display of collected digits
You can define when the system clears the display of collected digits (Callr-info) from the agent
telephone. The system allows the:
l
existing default option to clear the display when the next call is received
l
option to clear the display when the call is released
l
option to keep the displayed digits while the agent is in After Call Work (ACW) mode.
62 Avaya AuraTM Communication Manager Overview
CMS measurement of ATM
CMS measurement of ATM
The Call Management System (CMS) measurement of ATM feature provides the capability to
externally measure ATM trunks on CMS. The CMS messages and reports are modified to
support the expanded equipment location.
Dialed Number Identification Service
This feature displays, for a called party or answering position, the service or product associated
with an incoming call. You administer what the system displays.
Direct agent calling
Direct agent calling lets the customer's callers automatically go directly to the same agent
whenever they call for prompt, personalized service. These direct-to-the-agent calls are also
included in their call center measurement statistics.
Dual links to CMS
The dual links to CMS feature provides an additional TCP/IP link to a separate CMS for full,
duplicated CMS data collection functionality and high availability CMS configuration. The same
data is sent to both servers, and the administration can be done from either server.
The ACD data is delivered over different network routes to prevent any data loss from such
conditions as:
l
ACD link failures
l
CMS hardware or software failures
l
CMS maintenance
l
CMS upgrades
Issue 6 May 2009
63
Call Center
Duplicate agent login ID administration
Duplicate agent login ID administration simplifies administration of similar agent login ID forms.
Agent-loginID skill pair increase
Since the LINUX platform supports 20,000 administered agent-loginIDs, the administered
agent-loginID skill pairs has been increased from 65,000 to 180,000.
With this enhancement, customers could administer an average of 9 skills per agent for the
20,000 agent-loginIDs (180,000/20,000). Customers could also administer 9,000 agents with 20
skills each (180,000/20). The number of skill pairs is administered on the Display Capacity
SAT screen using the Administered Logical Agent-Skill Pairs field.
Note:
Note:
This capacity increase applies only to the S87XX Server and other configurations
that have the S87XX capacities.
Expert Agent Selection
Expert Agent Selection (EAS) enables certain skill types to be assigned to a call type or a
Vector Directory Number (VDN). Routing calls through vectoring then allows the system
administration to direct calls to agents who have the particular agent skills required to complete
the customer inquiries.
Add/remove skills
Allows an agent using expert agent selection (EAS) to add or remove skills. A skill is a numeric
identifier that refers to the specific ability of an agent. For example, an agent who speaks
English and Spanish could be assigned a language-speaking skill with an identifier of 20. The
agent then adds skill 20 to his or her set of working skills. If a customer needs a
Spanish-speaking agent, the system routes the call to an agent with that skill. Each agent can
have up to four active skills, and each skill is assigned a priority level.
64 Avaya AuraTM Communication Manager Overview
Least Occupied Agent
Call distribution based on skill
Calls that require certain agent skills (such as “knowledgeable about product X” or “speaks
Spanish”) can be matched to an agent who matches the required skill. You can assign one of up
to 999 skill numbers to each need or group of needs. The skills are administered and
associated for each of the following:
l
Vector directory numbers (VDN)
l
Agent login IDs
l
Callers
This refined skill definition capability allows you to organize call handling based on customer,
product, and language, for example.
Queue to best ISDN support
Queue to best information is passed transparently over several public networks and QSIG
private networks using the envelopes that are part of the QSIG Manufacturer-Specific
Information (MSI) and the ISDN platform enhancement.
Least Occupied Agent
The Least Occupied Agent (LOA) feature distributes calls evenly across all available agents,
balancing the workload among agents with fewer skills and agents with several skills. LOA
solves the problem of agents who are bombarded with calls after logging into a skill at the start
of a shift, while the agents who are already logged in have maintained their current incoming
call level.
Multiple call handling (forced)
This feature allows agents to receive an ACD call while other types of calls are alerting, active,
or on hold.
Issue 6 May 2009
65
Call Center
Multiple music/audio sources
Multiple music/audio sources lets customers deliver music or customized announcements to
callers while they are in queue, helping to make the waiting time more productive or
entertaining. Customers can provide information about their products, services, other call center
applications, offer public service information, or play music.
Locally sourced announcements and music
Use the Locally Sourced Announcements and Music feature to access announcement and
music audio sources on a local port network or media gateway.
Locally sourced audio can:
l
improve the quality of audio
l
reduce resource usage, such as VoIP resources
l
provide a backup mechanism for announcement and music sources
Multiple split queuing
Multiple split queuing lets customers direct a call to several splits at the same time, so that the
first available agent can take the call. It can help customers handle the busiest periods with
greater ease and provide faster service to their callers.
Network Call Redirection
Today, call center customers are looking for many ways to reduce their costs. One of these
ways is to employ Public Switched Telephone Network (PSTN) virtual private networks (VPNs)
to eliminate as much private network cost as possible. These cost reductions are particularly
valuable in enterprises or multi-site call-center environments and especially to enterprise call
centers where network costs are typically high.
Network call redirection (NCR) offers a call redirection method between sites on a public
network or a PSTN VPN, to help reduce trunking costs. NCR may only be activated for
incoming ISDN trunk calls where the associated trunk group has been enabled by the public
network service provider to use network call transfer or network call deflection features.
66 Avaya AuraTM Communication Manager Overview
PC Application Software Translation Exchange
ETSI Explicit Call Transfer signaling
The Network Call Redirection (NCR) support of the “ETSI Explicit Call Transfer” feature is
desired by multi-site, non-U.S. Avaya call center customers who use various PSTN service
providers for ISDN services. These non-U.S. call centers wish to accomplish call transfers
between sites without holding the ISDN trunks of a transferred call at the call redirecting
Communication Manager site.
The Network Call Redirection/Network Call Deflection (NCR/NRD) feature does not allow for
announcement and call-prompting call-vectoring operations. Therefore, the ETSI ECT feature is
for these call center customers who cannot use NCR/NRD since they wish to play an
announcement to a caller and use Communication Manager call-prompting to allow the caller to
determine the routing for the call.
Network call redirection 2B-channel transfer
This enhancement adds support for the 2B-Channel Transfer PSTN network transfer protocols
to the Network Call Redirection (NCR) feature. The protocols that are supported are:
l
l
Telcordia TBCT (offered by local and inter-exchange PSTNs with Lucent 5Ess or Nortel
DMS100 switches in US or Canada)
1998 ANSI Explicit Call Transfer (ECT) for future use.
Another form of network transfer is where the PBX sets up the second leg call and asks the
network to merge the incoming call with the outgoing call (the 2B-channels) and drops the
trunks to the PBX.
PC Application Software Translation Exchange
PC Application Software Translation Exchange (PASTE) allows users to view call center data
on display telephones, displaying what each terminal button is, and what the feature access
codes for the switch are. PASTE is used in conjunction with Avaya IP agent.
Priority queuing
Priority queuing allows special callers to be assigned priority status and routed ahead of other
callers. Clients can pamper their best customers with the fastest attention possible.
Issue 6 May 2009
67
Call Center
Reason codes
Allows agents to enter a numeric code that describes their reason for entering auxiliary (AUX)
work mode or for logging out of the system. Reason codes give call center managers detailed
information about how agents spend their time. You can use this data to develop more precise
staffing forecasting models or use it with schedule-adherence packages to ensure that agents
are performing scheduled activities at the scheduled time. You must have expert agent
selection (EAS) enabled to use reason codes.
Redirection on no answer
This feature redirects a ringing ACD split or skill call or direct agent call after an administered
number of rings. This prevents an unanswered call from ringing indefinitely. The call can redirect
either to the split or skill to be answered by another agent or to a Vector Directory Number
(VDN) for alternative call handling. Direct agent calls route to the agent coverage path, or to a
VDN if no coverage path is administered. You must have ACD enabled to use this feature.
Remote logout of agent
The remote logout of agent feature allows a select set of users to log out an agent using a
feature access code.
Service observing
Service observing allows a specified user, such as a supervisor, to observe or monitor calls of
another user. A vector directory number call can also be observed. Observers can observe in
listen-only or listen-and-talk mode. You set up service observing to observe a particular
extension, not all calls to all extensions at a terminal.
Note:
Note:
Service observing may be subject to federal, state, or local laws, rules, or
regulations or require the consent of one or both of the call parties. Familiarize
yourself and comply with all applicable laws, rules, and regulations before using
this feature.
68 Avaya AuraTM Communication Manager Overview
Service observing
Listen-only FAC for service observing
The system provides a no-talk, listen-only service observing feature access code (FAC). This
FAC does not reserve a second timeslot for potential toggle to talk and listen mode. This feature
is for call recording applications that use Service Observing of stations/ACD agents to provide
increased call recording capacity by reducing the timeslot usage.
Note:
This capability only applies to Port Network Gateways such as the G650. The
additional timeslot is still reserved in H.248 Media Gateways.
Note:
Service observing by COR
Service observing by class of restriction (COR) restricts certain users from using the service
observing feature.
Service observing of VDNs
Service observing of VDNs (also known as VDN observing on agent answer) allows a
supervisor to start observing a call to the VDN when the call is delivered to the agent station.
The observer will not hear the call during vector processing (announcements, music, and so
on).
Service observing remote
This option will allow observing from non-feature button equipped stations. An observer will be
able to monitor a VDN or a physical extension remotely using an “observe FAC” procedure
through the remote access feature and/or call vectoring/call prompting features (through VDNs).
Service Observing with Multiple Observers
Up to two observers can monitor the same agent Login ID or station extension using the Service
Observing station button or using any of the following Feature Access Codes (FACs):
l
Service Observing Listen-Only
l
Service Observing Listen/Talk
l
Service Observing No-Talk
Issue 6 May 2009
69
Call Center
Two separate calls, each with an associated service observer, can be conferenced together with
both service observers included in the merged conferenced call except when both observers
are VDN observers. In this case one VDN observer will be dropped.
Customers who use call recording products, such as the Avaya Witness Call Recording or NICE
can connect a voice-storage server to a station or Login ID extension in order to record
agent-to-customer transactions acting as an observer. Call recording observing can be given
priority.
Customers who use call recording products can also allow an observer to monitor a station or
Login ID extension and record the transaction at the same time.
Note:
Note:
This feature does not allow multiple observers on the same call for the Service
Observing by VDN feature.
Vector-initiated service observing
Vector-initiated service observing, also called VDN observing on agent answer, allows users to
start observing of a call to the VDN when the call is delivered to the agent or station. This saves
time for the observer after observing of the VDN has been activated since the observer does not
have to wait listening for each subsequent call to go through vector processing and for the
agent to answer.
Site statistics for remote port networks
The site statistics for remote port networks feature forwards location IDs to CMS to provide call
center site-specific reports.
User-to-user information over the public network
This feature provides the mechanism to pass information across several key public networks,
including information that is originated or destined for one of several applications on
Communication Manager.
70 Avaya AuraTM Communication Manager Overview
Voice Response Integration
Voice Response Integration
Voice Response Integration (VRI) integrates call vectoring with the capabilities of voice
response units such as the Avaya CONVERSANT voice information system. You can also
integrate a voice response unit with ACD. All this provides a variety of advantages. For
example, while a call is queued, a caller can listen to product information via an audiotext
application or can complete an interactive voice-response transaction. It may be possible to
resolve the caller questions while the call is queued, which helps reduce queuing time for other
callers during peak times.
VuStats
VuStats presents BCMS statistics on telephone displays. Agents, supervisors, call center
managers, and other users can press a button and view statistics for agents, splits or skills,
VDNs, and trunk groups. These statistics can help agents monitor their own performance, or
respond appropriately to the caller request. Features include:
l
VuStats login IDs
l
VuStats service level
Issue 6 May 2009
71
Call Center
72 Avaya AuraTM Communication Manager Overview
Chapter 5: Collaboration
Communication Manager contains a variety of features aimed at providing easy ways to
collaborate with groups of peers, customers, and partners such as executives, sales people,
and professional specialists. These key work groups require a high level of effective interaction,
and Communication Manager delivers.
This chapter is divided into three sections:
l
Conferencing
l
Multimedia calling
l
Paging and intercom
Conferencing
Abort conference on hang-up
When you press the conference button and for any reason you hang up before you complete
the conference, you will cancel the conference. The original call that was put on soft-hold is put
on hard-hold.
Conference - three party
The conference button allows single-line telephone users to make up to three-party conference
calls without attendant assistance.
Conference - six party
The conference button allows multi-appearance telephone users to make up to six-party
conference calls without attendant assistance.
Issue 6 May 2009
73
Collaboration
Conference/transfer display prompts
Conference/transfer display prompts are based on the user class of restriction (COR). The
display prompts are based on the user COR, independent of the select line appearance
conferencing and no dial tone conferencing feature. The display messages vary depending on
the activation of the two features, but the choice of displaying the additional information or not is
dependent on the station user COR.
Conference/transfer toggle/swap
The conference/transfer toggle/swap feature allows users to toggle between two parties in the
middle of setting up a conference call prior to connecting all parties together, or to consult with
both parties prior to transferring a call. The display also toggles between the two parties.
Group listen
The group listen feature simultaneously activates your speakerphone in listen-only mode, and
your handset or headset in listen-and-speak mode. This allows you to serve as spokesperson
for a group. You can participate in a conversation while everyone else in the room is listening to
what is said.
Note:
Note:
This feature works only on certain types of telephones. It is not supported on IP
telephones.
Hold/unhold conference
Allows user to use the Hold button to bring the held party back to the conversation. This is an
alternative to using the line appearance button. Hold/unhold only applies if there is only one line
on hold and no other line appearances are active. An error message is displayed if the unhold
feature is attempted when not allowed.
Note:
Note:
This feature is not available for BRI stations or attendant consoles.
74 Avaya AuraTM Communication Manager Overview
Conferencing
Meet-me Conferencing
The Meet-me Conferencing feature allows a person to set up a dial-in conference of up to six
parties. The Meet-me Conferencing feature uses call vectoring to process the setup of the
conference call.
Meet-me Conferencing can be optionally set up to require an access code. If an access code is
assigned, and if the vector is programmed to expect an access code, each user dialing in to the
conference call must enter the correct access code to be added to the call.
The Meet-me Conferencing extension can be dialed by any internal or remote access users,
and by external parties if the extension number is part of the customer DID block.
Expanded Meet-me Conferencing
Use the Expanded Meet-me Conferencing application to set up multi-party conferences
consisting of more than six parties. The Expanded Meet-me Conferencing application supports
up to 300 parties. This application is available with Communication Manager release 3.0 or
later.
The Expanded Meet-me Conferencing application requires an external Meeting Exchange (MX)
server. For more information, see the Expanded Meet-me Conferencing section of the Avaya
Aura™ Communication Manager Feature Description and Implementation, 555-245-205. Also
see the MX server documentation.
No dial tone conferencing
This feature can eliminate user confusion over receiving dial tone when trying to conference two
existing calls. It skips the automatic line selection if there is already a party on hold or an alerting
line appearance. Help messages help guide the user. This feature is assigned on a system wide
basis.
No hold conference
This feature allows a user to automatically add another party to a conference call while
continuing the conversation of the existing call. The new party is automatically entered into the
conversation as soon as the call is answered. An optional tone can be provided prior to the
party being added to the call.
Issue 6 May 2009
75
Collaboration
Note:
The calling station cannot hold, conference, or transfer an Emergency Access to
Attendant call. This applies to both the traditional means of using these features,
and to the no-hold method of using these features.
Note:
After dialing is complete, if the No Hold Conference is not answered within the time specified in
an administered “timeout” field, the No Hold Conference call is deactivated.
Select line appearance conferencing
If you are in a conversation on line “b”, and another line is on hold or an incoming call is alerting
on line “a”, then pressing the CONF button bridges the calls together. Using the select line
appearance feature on Communication Manager, the user has the option of pressing a line
appearance button to complete a conference instead of pressing CONF a second time.
This feature only applies if the line is placed in soft hold by pressing the CONF button. This
feature never applies if the soft hold was due to pressing a TRANSFER button.
Selective conference party display, drop, and mute
The selective conference party display, drop, and mute feature allows any user on a digital
station with display or on an attendant console to use the display to identify all of the other
parties on a two-party or conference call.
The user would press a feature button while on the call that puts the station or console into
conference display mode. The user then can scroll through the display of each party currently
on the call by repeatedly pressing the feature button. The display would show the number and
name (when available) of the caller.
The user could then do either of the following:
l
l
The user can selectively drop the party currently shown on the display with a single button
push. This can be useful during conference calls when adding a party that does not
answer and the call goes to voice mail.
The user can selectively mute the party currently shown on the display with a single button
push. This puts the selected party in “listen-only” mode. This can be useful during
conference calls when a party puts the conference call on hold and everyone on the call is
forced to listen to music-on-hold. The user can mute that party so the conference call can
continue without interruption. The muted party can then rejoin the call by pressing the #
key on their telephone.
76 Avaya AuraTM Communication Manager Overview
Multimedia calling
! CAUTION:
Station users must be careful when scrolling through the displays when using the
selective conference party display feature. The station hyperactivity feature will
take the station out of service if the user repeatedly scrolls through the displays at
high enough rates. This causes the station to be reset and the user is dropped
from the call.
CAUTION:
The Expanded Meet-me Conferencing application impacts selective display for all types of
conferences. For more information, see the Expanded Meet-me Conferencing section of the
Avaya Aura™ Communication Manager Feature Description and Implementation, 555-245-205.
Selective conference mute
Selective conference mute allows a conference call participant, who has a display station, to
mute a noisy trunk line. Selective conference mute is also known as far end mute.
Examples of noisy trunk lines that might need to be muted during a conference call are:
l
cell telephones
l
telephones that utilize the Music-On-Hold feature
l
telephones with no mute capabilities
Selective conference mute only applies to trunk lines on the conference call, and not to stations.
Only one trunk line on the conference call can be selectively muted at a time. This enhanced
conferencing feature can be activated from any display station with a “conf-dsp” button and an
“fe-mute” button.
The selective conference mute feature works with any conference established through
Communication Manager, either a traditional 3 or 6 party conference or a Meet-Me conference.
Note:
Note:
This feature requires that the enhanced conferencing feature be set to Y on the
“system-parameters customer-options” screen.
Multimedia calling
Multimedia calls are initiated with voice and video only. Once a call is established, one of the
parties may initiate an associated data conference to include all of the parties on the call who
are capable of supporting data. The data conference is controlled by an adjunct device called
an Expansion Services Module (ESM).
Issue 6 May 2009
77
Collaboration
Multimedia Application Server Interface
The multimedia Application Server Interface (ASA) provides a link between Communication
Manager and one or more multimedia communications eXchange nodes. A Multimedia
Communications Exchange (MMCX) is a stand-alone multimedia call processor produced by
Avaya. This link to Communication Manager enhances the capabilities of each multimedia
communications eXchange system by enabling it to share some of the Communication
Manager features.
In particular, the interface provides the following advantages:
l
l
l
Call Detail Recording (CDR) - This allows you to capture call detail records so you can
analyze the call patterns and usage of multimedia calls just as Communication Manager
administrators analyze normal calls.
Automatic Alternate Routing/Automatic Route Selection (AAR/ARS) - This allows for the
intelligent selection of the most cost-effective routing for calls, based on available
resources and your carrier preference. The system may select public trunks through a
DEFINITY® MultiMedia Communication Exchange (MMCX).
Voice mail integration - You can access your embedded AUDIX or INTUITY AUDIX voice
messaging system from a MultiMedia Communication Exchange (MMCX).
Multimedia call early answer on vectors and stations
Early answer is a feature applied to multimedia calls in conjunction with conversion to voice.
The early answer feature:
l
Answers the data call
l
Establishes the multimedia protocol prior to completion of a converted call
l
Ensures that a voice path to/from the originator is available when the voice call is
answered
For an incoming call, early answer answers the dynamic service-link calls when the destination
endpoint answers, unless early answer is specified during routing or termination processing.
Note:
Note:
The “destination voice endpoint” might be an outgoing voice trunk if the
destination voice station is forwarded or covered off-premises.
Multimedia Call Handling
See Multimedia Call Handling on page 178.
78 Avaya AuraTM Communication Manager Overview
Paging and intercom
Multimedia call redirection to multimedia endpoint
A dual port multimedia station may be a destination of call redirection features such as call
coverage, forwarding, and station hunting. The station can receive and accept full multimedia
calls or data calls converted to multimedia.
Multimedia data conferencing (T.120) through an ESM
The data conference is controlled by an adjunct device called an Expansion Services Module
(ESM). The ESM is used to terminate T.120 protocols [including Generalized Conference Call
(GCC), a protocol standard for data conference control] and provide data conference control
and data distribution. The MultiMedia Interface circuit pack, TN787, is used to rate adapt T.120
data to/from the ESM.
For more information on ESM, see Installation for Adjuncts and Peripherals for Avaya Aura™
Communication Manager .
Multimedia hold, conference, transfer, and drop
Station users have the ability to activate hold, conference, transfer, or drop on multimedia calls.
Multimedia endpoints and voice-only stations may participate in the same conference.
Multimedia queuing with voice announcement
When multimedia callers queue for an available member of a hunt group, they are able to hear
an audio announcement.
Paging and intercom
Code calling access
This feature allows attendants, users, and tie trunk users to page with coded chime signals.
This feature is helpful for users who are often away from their telephones or at a location where
a ringing telephone might be disturbing.
Issue 6 May 2009
79
Collaboration
Group paging
Group paging allows a user to make an announcement to a group of people using
speakerphones. The speakerphones are automatically turned on when the user begins the
announcement. The recipients can listen to the message over the handset if they wish, but they
cannot speak to the user in return.
A group page member will not receive the page if the member is active on a call appearance,
has a call ringing, is off-hook, has “send-all calls” active, or has “do not disturb” active.
Intercom - automatic
With this feature, users who frequently call each other can do so by pressing one button instead
of dialing an extension number. Calling users press the automatic intercom button and lift the
handset. The called user receives a unique intercom ring and the intercom lamp, if provided,
flashes.
Intercom - automatic answer
Automatic answer intercom (auto answer ICOM) allows a user to answer an intercom call within
the intercom group without pressing the intercom button. Auto answer ICOM works with digital,
BRI, and hybrid telephones with built-in speaker, headphones, or adjunct speakerphone.
Intercom - dial
This feature allows multi-appearance telephone users to easily call others within an
administered group. The calling user lifts the handset, presses the dial intercom button, and
dials the one-digit or two-digit code assigned to the desired party. The telephone of the called
user rings, and the intercom lamp, if provided, flashes. With this feature, a group of users who
frequently call each other can do so by pressing one button and dialing a one-digit or two-digit
code instead of dialing an extension number.
80 Avaya AuraTM Communication Manager Overview
Paging and intercom
Loudspeaker paging access
Loudspeaker paging access provides attendants and telephone users dial access to voice
paging equipment. As many as nine paging zones can be provided by the system, and one
zone can be provided that activates all zones at the same time.
Note:
Note:
A zone is the location of the loudspeakers - for example, conference rooms,
warehouses, or storerooms.
A user can activate this feature by dialing the trunk access code of the desired paging zone, or
the access codes can be entered into abbreviated dialing lists. Once you have activated this
feature, you can simply speak into the handset to make the announcement.
Deluxe loudspeaker paging access (called deluxe paging) provides attendants and telephone
users with integrated access to voice-paging equipment and call park capabilities. When you
activate deluxe paging, the call is automatically parked. The parked call returns to the parking
user with distinctive alerting when the time-out interval expires.
Manual signaling
Allows one user to signal another user. The receiving user hears a two-second ring. The signal
is sent each time the button is pressed by the signaling user. The meaning of the signal is
prearranged between the sender and the receiver. Manual signaling is denied if the receiving
telephone is already ringing from an incoming call.
Whisper page
Whisper page allows an assistant or colleague to bridge onto your telephone conversation and
give you a message without being heard by the other party or parties you are talking to. Whisper
page works only on certain types of telephones.
Issue 6 May 2009
81
Collaboration
82 Avaya AuraTM Communication Manager Overview
Chapter 6: Communication device support
Avaya IP Agent
Avaya IP Agent is a PC-based IP application that allows agents to use their PCs as telephones.
In addition to the traditional functionality of a standard telephone (transfer, hold, conference,
and so forth), IP agent offers directory services, screen pops, call history, and agent mode
history.
Avaya IP Softphone
Avaya IP Softphone extends the level of Communication Manager services. This feature turns a
PC or a laptop into an advanced telephone. Users can place calls, take calls, and handle
multiple calls on their PCs.
Note:
R1 and R2 IP Softphone and IP Agent, which use a dual connect (two
extensions) architecture, are no longer supported. R3 and R4 IP Softphone and
IP Agent, which use a single connect (one extension) architecture, continue to be
supported. This applies to the RoadWarrior configuration and the Native H.323
configuration for the IP Softphone.
Note:
The R5 release of the IP Softphone supports a number of enhanced features, including the
following:
l
Improved endpoint connection recovery algorithm
l
AES media encryption (see Encryption algorithm for bearer channels on page 200)
l
Instant Messaging
l
Unicode support (see Unicode support on page 87)
l
Softphone and Telephone Shared Control (see IP Softphone and IP Agent - Shared
Control mode on page 84)
The IP Softphone provides a graphical user interface with enhanced capabilities when used
with certain models of DCP telephones. Communication Manager supports a mode of H.323
registration that allows an IP Softphone to register for the same extension as a DCP telephone
without disabling the telephone. It also allows the IP Softphone to send button-push messages
and receive display and call progress messages in parallel with the telephone. In this mode, the
Softphone does not terminate any audio.
Issue 6 May 2009
83
Communication device support
IP Softphone and IP Agent - RoadWarrior mode
IP Softphone and IP Agent, RoadWarrior mode, enables use of the full Avaya Communication
Manager feature set from temporary remote locations anywhere in the world. The RoadWarrior
application consists of two software applications running on a PC that is connected to
Communication Manager over an IP network.
The single network connection between the PC and Communication Manager carries two
channels, one for the signaling path and one for the voice path. On Communication Manager,
the RoadWarrior application requires the CLAN circuit pack for signaling and the IP media
processor for voice processing.
IP Softphone and IP Agent - Shared Control mode
IP Softphone and IP Agent, Shared Control mode, enables users to have a telephone endpoint
and an IP Softphone in service simultaneously on the same extension number. IP Softphone
and an IP telephone can be integrated so that the IP softphone can control a desk IP telephone.
This allows the power of the PC desktop (LDAP directories, TAPI PIMs/Contact Managers, etc.)
to be used in conjunction with a desktop IP telephone.
An IP softphone can register to an extension number that is already assigned to an in-service
telephone endpoint. From that point on, user actions carried out by either endpoint apply to calls
to or from the extension. Only the telephone endpoint carries audio for the extension, however.
IP Softphone and IP Agent - Telecommuter mode
IP Softphone and IP Agent, Telecommuter mode, enables telecommuters to use the full
Communication Manager feature set from home. It consists of a PC and a telephone with
separate connections to Communication Manager. The PC provides the signaling path and the
user interface for call control. A standard telephone provides a high-quality voice path. The
Telecommuter application requires the CLAN circuit pack for signaling. The Telecommuter
application does not use the IP media processor.
Avaya IP Softphone for pocket PC
Avaya IP Softphone for pocket PC extends the level of Communication Manager services. This
feature turns a hand-held personal digital assistant (PDA) into an advanced telephone. Users
can place calls, take calls, and handle multiple calls on their PDAs.
84 Avaya AuraTM Communication Manager Overview
Communication Manager PC console
Communication Manager PC console
The Communication Manager PC console allows your attendants to efficiently handle incoming
calls by personal computer. Using the familiar Microsoft Windows graphical user interface
(GUI), the attendants can easily keep track of how long callers have been on hold and who they
are waiting for. Attendants can monitor up to six calls at once.
Attendants do not need to use pen and paper when handling calls because they can make
notes on their computers about what each caller needs. All this contributes to make a favorable
first impression with your customers. Having the call processing software on the same computer
with spreadsheet, word processing, or other software allows the attendants to stay productive
between calls.
The PC console is easily customized, so even if attendants from different shifts share the same
computer, they can each preserve their preferences in the call processing environment. The PC
console is available in English, Parisian French, Latin American Spanish, German, Dutch,
Italian, and Portuguese. If a Spanish-speaking attendant takes over for a French-speaking
attendant, for example, a single press of a button converts all labels, error messages, and
online help to Spanish.
Avaya one-X Communicator
Avaya one-X Communicator is Avaya's new Unified Communications client that provides
enterprise users with simpler, more intuitive access to all their everyday communications tools.
The initial release supports both SIP and H.323 networks as well as rich presence and video to
enhance collaboration and face-to-face communications. Avaya one-X Communicator can be
deployed as a standalone client for softphone capabilities with or without video or can be
integrated with the Avaya one-X Portal Server for a richer set of unified features.
Avaya one-X Portal as software-only phone
Avaya one-X Portal is a browser based interface to Avaya telephony, messaging, mobility,
conferencing, and presence services provided by Communication Manager, Avaya Modular
Messaging, Avaya Meeting Exchange, and Avaya Intelligent Presence Server. Avaya one-X
Portal does not require the installation of any application software on your desktop to deliver its
basic functionality.
Avaya one-X Portal provides the following features:
l
Single web client interface
Issue 6 May 2009
85
Communication device support
l
Communication Manager telephony features
l
Any telephone can access Communication Manager features
l
Telephony control with supported versions of Communication Manager installed in your
enterprise
l
Customizable call logs
l
Integration with Avaya Modular Messaging to view and play voice messages
l
Integration with Meeting Exchange to view and control live conferences
l
l
l
Integration with Avaya Intelligent Presence Server to receive access requests and publish
presence state information.
Integration with Extension to Cellular for Follow-Me applications
Integration with Microsoft Active Directory, IBM Domino Server, Novell eDirectory, or Sun
One Directory Server for enterprise user information
Avaya SIP softphone
Avaya SIP Softphone is a client-based SIP application for the PC or laptop running the
Microsoft Windows operating system. Avaya SIP Softphone supports Road Warrior mode, uses
the SIP protocol to allow users to make and receive telephone calls, send and receive instant
messages, and see enterprise contact availability via presence.
Avaya SoftConsole
The Avaya SoftConsole is a Windows-based GUI application that can replace the physical 302B
“hard” console. It allows attendants to perform call answering and routing through a PC
interface through an IP connection.
Avaya SoftConsole - RoadWarrior mode
Avaya SoftConsole, RoadWarrior mode, enables use of the full Avaya Communication Manager
feature set from temporary remote locations anywhere in the world. The RoadWarrior
application consists of two software applications running on a PC that is connected to
Communication Manager over an IP network.
The single network connection between the PC and Communication Manager carries two
channels, one for the signaling path and one for the voice path. On Communication Manager,
86 Avaya AuraTM Communication Manager Overview
Increased text field length for feature buttons - DCP
the RoadWarrior application requires the CLAN circuit pack for signaling and the IP media
processor for voice processing.
Avaya SoftConsole - Telecommuter mode
Avaya SoftConsole, Telecommuter mode, enables telecommuters to use the full
Communication Manager feature set from home. It consists of a PC and a telephone with
separate connections to Communication Manager. The PC provides the signaling path and the
user interface for call control. A standard telephone provides a high-quality voice path. The
Telecommuter application requires the CLAN circuit pack for signaling. The Telecommuter
application does not use the IP media processor.
In telecommuter mode, both the SoftConsole and the physical handset telephone must be
answered to answer the call.
Increased text field length for feature buttons - DCP
The increased text field length for feature buttons - DCP feature is designed to allow the switch
to better support the longer text label display capabilities that are available on the newer DCP
(24xx) model telephones.
This feature allows the end user to program and store 13-character labels for all feature buttons
and call appearances associated with the DCP telephones on the switch. The only exceptions
are those labels that are administratively blocked or disabled.
Unicode support
Communication Manager supports the display of non-English static and dynamic display text on
Unicode-enabled telephones. Non-English display information is entered into a Avaya
Integrated Management application. Communication Manager processes, stores, and transmits
the non-English text to telephones that support Unicode displays.
Unicode support provides the capability of supporting international and multi-national
communications solutions. End-users are provided with a communications interface (delivered
by an IP telephone or IP Softphone) in their own native language. This feature supports the
Simplified Chinese, Japanese, and Korean (CJK) character sets.
Issue 6 May 2009
87
Communication device support
QSIG support for Unicode
The QSIG support for Unicode feature extends the Unicode support on a single server to
multi-node Communication Manager networks. This feature allows Unicode support across
large campus configurations. Many configurations contain multiple Communication Manager
servers due to scalability requirements. This feature also allows Unicode support across large
corporate networks, frequently multinational corporations, where multiple Communication
Manager servers are almost always provisioned.
88 Avaya AuraTM Communication Manager Overview
Chapter 7: Hospitality
Alphanumeric dialing
Alphanumeric dialing allows you to place data calls by entering an alphanumeric name rather
than a long string of numbers.
Attendant room status
See Attendant room status on page 38.
Automatic selection of Direct Inward Dialing numbers
This feature allows the system to automatically choose a number from a list of available Direct
Inward Dialing (DID) numbers that will be assigned to a guest room extension when checking in.
With this feature, hotels can give a guest a second telephone number that is different from their
room number, thereby protecting the privacy of the guest. When a particular DID number is
called, the call routes to the guest room extension, and covers as if the room was called directly.
Besides improving guest security, this eliminates the need for an attendant or front desk staff to
extend a call to a guest room.
Automatic wakeup
The automatic wakeup feature allows attendants, front desk users, and guests to request that
one or two wake-up calls be automatically placed to a certain extension number at a later time.
When a wakeup call is placed and answered, the system can provide a recorded
announcement (which can be a speech synthesis announcement), music, or simply silence.
With the integrated announcement feature, multiple announcements enable international guests
to use wakeup announcements in a variety of languages. See Daily wakeup on page 90, Dual
wakeup on page 91, and VIP wakeup on page 93.
Issue 6 May 2009
89
Hospitality
Check-in/check-out
This feature allows front desk personnel to check guests into a hotel and, when the guests
leave, check them out. There are two ways this is done: through the PMS terminal or through
the attendant console (or backup telephone). Check-in and check-out from the attendant
console should be used only if there is no Property Management System (PMS), or if the link to
the PMS is down. If the PMS is installed and working, check guests in and out using the PMS.
For guest check-in or check-out from the console, there are two buttons on the attendant
console (or backup telephone): one labeled “Check in” and the other labeled “Check out.” The
check-in procedure performs two functions: it deactivates the restriction on the telephone in the
room allowing outward calls, and it changes the status of the room to occupied.
Custom selection of VIP DID numbers
This feature builds on the automatic selection of DID numbers feature. It allows hotel personnel
to control what DID number is assigned to a hotel room at check-in. That is, the system asks the
user to specify the desired DID number when a guest is checked in. The number comes from a
pool of DID numbers that are separate from those used by the automatic selection feature. The
system never automatically assigns numbers from this pool. Numbers from this pool are used
only when explicitly specified by the user.
Daily wakeup
Daily wakeup allows a guest or front desk personnel to schedule a single wakeup request for a
daily wakeup call. For example, if a guest needs to receive a wakeup call at 5:30 a.m. for the
duration of his or her stay, one request can be placed on the system instead of placing a
separate request for each day.
Dial-by-name
The dial-by-name feature allows callers to the system to access guest rooms simply by dialing
the name of the guest they are trying to contact. This feature uses recorded announcements
and the call vectoring feature to set up an automatic attendant procedure.
90 Avaya AuraTM Communication Manager Overview
Do not disturb
This automatic attendant procedure gives callers the ability to enter a guest name. When a
single or unique match is found, the call is redirected to the telephone of the guest.
Do not disturb
The do not disturb feature allows guests, attendants, and authorized front desk users to request
that no calls, other than priority calls, be connected to a particular extension until a specified
time.
Dual wakeup
This feature allows guests to have two separate wakeup calls. The dual wakeup feature is an
enhancement to the standard automatic wakeup feature used in hospitality environments.
With the standard wakeup feature, guests or front desk personnel can create one wakeup call
for each extension. The dual wakeup feature allows guests and front desk personnel to create
either one or two wakeup calls. The dual wakeup feature for guests is valid only when the
system is not equipped with a speech synthesizer circuit pack.
Housekeeping status
The housekeeping status feature records the status for up to six housekeeping codes and
reports them to the property management system (PMS). These status codes are usually
entered by the housekeeping staff from the guest room or from a designated telephone. They
can also be updated by the front office personnel using the attendant console or a backup
telephone. Six status codes can be used from guest rooms, and four status codes can be used
from telephones that do not have the client room class of service (COS).
Names registration
The names registration feature automatically sends a guest name and room extension from the
property management system (PMS) to the switch at check-in, and automatically removes this
information at check-out. The information may be displayed on any attendant console or
display-equipped telephone at various hotel locations (for example, room service or security).
Issue 6 May 2009
91
Hospitality
Property Management System digit to insert/delete
Many customer configurations base a room extension by adding an extra leading digit on the
room number. The PMS digit to insert/delete feature allows users to delete the leading digit of
the extension in messages. The feature is useful for a hotel that has multiple extensions sharing
an extra leading digit in front of the room number. The leading digit is automatically inserted
when the message goes to the switch.
The PMS interface supports 3-digit, 4-digit, or 5-digit extensions, but prefixed extensions do not
send the entire number across the interface. Only the assigned extension number is sent.
Therefore, you should not use prefixed extensions for numbers that are also going to use the
digit to insert/delete function.
Property Management System interface
The Property Management System (PMS) allows a customer to control features used in both a
hospital-type and a hotel/motel-type environment. The communications link allows the property
management system to interrogate the switch, and allows information to be passed between the
switch and the PMS. The switch exchanges guest status information (room number, call
coverage path, and other data) with the PMS.
There are two ways that the guest data can be encoded:
l
l
Using a combination of Binary Coded Decimal (BCD) encoding and the ASCII character
set
Using only the ASCII character set
Single-digit dialing and mixed station numbering
This feature provides hotel staff and guests easy access to internal hotel/motel services, and
provides the capability to associate room numbers with guest room telephones. The feature
provides the following dial plan types: single-digit dialing, prefixed extensions, and mixed
numbering.
92 Avaya AuraTM Communication Manager Overview
Suite check-in
Suite check-in
Suite check-in allows more than one station to be checked in at one time. This is useful for a
guest room that may have multiple extensions. This feature allows all extensions to be checked
in at the same time. Suite check-in using the hunt-to feature will also check out all the
extensions in the entire suite at the same time.
VIP wakeup
The VIP wakeup feature allows front desk personnel to provide personalized wakeup calls to
important guests. When a wakeup call has been scheduled for an important guest, a wakeup
reminder call is placed to the front desk personnel, who in turn personally calls the guest to
provide the wakeup call.
Wake-up activation using confirmation tones
If a speech synthesizer circuit pack is not installed, guests can still enter their own wakeup calls
(two wakeup calls if the dual wakeup feature is active). The guests do not receive voice prompts
as they would using the speech synthesizer circuit pack. Instead, guests receive call progress
tones (recall dial tone and confirmation tone) to set up their wakeup calls.
Xiox call accounting
The Xiox call accounting works as an adjunct with any system with hospitality features. Xiox call
accounting allows hotel management to use their telephone system as a major source of
revenue by generating the information they need to make important decisions about their
network and usage.
Issue 6 May 2009
93
Hospitality
94 Avaya AuraTM Communication Manager Overview
Chapter 8: Localization
Administrable language displays
This feature allows messages that appear on telephone display units to be shown in the
language spoken by the user. These messages are available in English (the default), French,
Italian, Spanish, or one other user-defined language. The language for display messages is
selected by each user. The feature requires 40-character display telephones.
The display messages are provided for standard telephone features such as Station Security
Code, Personal Station Access, Call Pickup, Station Lock, Extension to Cellular, Call
Forwarding, Call Coverage, and Call Park. This feature applies only to DCP and IP-H.323
terminals. This feature does not apply to SIP telephones and stations that act as attendant
consoles.
Administrable loss plan
The administrable loss plan provides the ability to administer signal loss and gain for telephone
calls. This capability is necessary because the amount of loss allowed on voice calls can vary
by country. With the administrable loss plan feature, switch endpoints are classified into 17
endpoint types, and the loss plan can be administered for trunks, stations, and personal CO
lines. Loss values are in the range of 15 dB loss to 3 dB gain. Preset defaults are available and
are based on country type.
Bellcore calling name ID
This feature allows the system to accept calling name information from a Local Exchange
Carrier (LEC) network that supports the Bellcore calling name specification. The system can
send calling name information in the format if Bellcore calling name ID is administered. The
following caller ID protocols are supported:
l
Bellcore (default) - US protocol (Bellcore transmission protocol with 212 modem protocol)
l
V23-Bell - Bahrain protocol (Bellcore transmission protocol with V.23 modem protocol).
Issue 6 May 2009
95
Localization
Block collect call
This feature blocks collect calls on class-of-restriction basis. This feature is available for any
switch that uses the Brazil country code. If enabled for a station, all trunk calls that terminate to
the station will send back a double answer to the central office (CO). This double answer tells
the CO that this particular station cannot accept collect calls. The CO then tears down the call if
it is a collect call.
Busy tone disconnect
In some regions of the world, the CO sends a busy tone for the disconnect message. With busy
tone disconnect, the switch disconnects analog loop-start CO trunks when a busy tone is sent
from the CO.
Country-specific localization
Italy
Distributed Communications Systems protocol
Enhanced DCS adds features to the existing DCS capabilities and requires the use of Italian
TGU/TGE tie trunks.
Additional features include:
l
Exchanging information to provide class of restriction (COR) checking between switches in
the EDCS network
l
Providing call-progress information for the attendant
l
Allowing attendant intrusion between a main and a satellite PBX
l
Allowing a main PBX to provide DID/CO intercept treatment rather than the satellite PBX
96 Avaya AuraTM Communication Manager Overview
Country-specific localization
Japan
National private networking support
Provides support for Japanese private ISDN networks. The Japanese private network ISDN
protocol is different from the standard ISDN protocol. The switch supports extensions to the
ISDN protocol for switches using the Japanese country code.
Katakana character set
Communication Manager supports the katakana character set (Japan). This nine-point
character font was designed to display katakana characters in the user interface as well as in
switch-generated messages.
Russia
Central Office support on G700 Media Gateway
Communication Manager supports central office (CO) trunks in Russia using the G700 Media
Gateway.
ISDN/DATS network support
This feature supports ISDN/DATS trunk networks when the tone generated field is set to
15 (Russia) on the system-parameters country-options screen. It modifies the overlap sending
delay and ISDN T302 and T304 timers to support the Russian trunk network.
Multi-Frequency Packet signaling
Multi-Frequency Packet (MFP) address signaling is provided in Russia on outgoing CO trunks.
Calling party number and dialed number information is sent on outgoing links between local and
toll switches. Russian MFP is set on each trunk group on the type field on the trunk screen.
Note:
Note:
Russian MFP does not apply to PCOL trunks.
Issue 6 May 2009
97
Localization
E&M signaling - continuous and pulsed
Continuous and pulsed E&M signaling is a modification to the E&M signaling used in the United
States. Continuous E&M signaling is intended for use in Brazil, but can also be used in Hungary.
Pulsed E&M signaling is intended for use in Brazil.
Multinational Locations
For customers who operate in more than one country, the Multinational Locations feature
provides the ability to use a single Enterprise Communication Server (ECS) across multiple
countries with:
l
telephones
l
port networks
l
remote offices
l
media gateways
The Multinational Locations feature allows the following Communication Manager features to
work across international borders:
l
A & Mu law companding
l
Call Progress Tone Generation
l
Loss Plan
l
Analog line board parameters
l
Call Detail Recording
l
R2-MFC (multifrequency signaling) trunks
The Multinational Locations feature works across all Linux platforms supported by
Communication Manager release 2.1 or higher.
The S8300, S8500, and S87XX Servers each supports 25 location parameter sets. You can
administer one parameter set for each country that you support, for a maximum of 25 countries.
Note:
Note:
Since the S8100 Server supports only 1 location, and since the Multinational
Locations feature depends on multiple locations, the Multinational Locations
feature is not supported on the S8100 platform.
98 Avaya AuraTM Communication Manager Overview
Multinational Locations
Analog line board parameters per location
You can administer the following analog line board parameters for each location:
l
Analog Ringing Cadence
l
Analog Line Transmission
l
Flashhook Interval Upper Bound
l
Flashhook Interval Lower Bound
l
Forward Disconnect Timer (msec)
l
Analog line tests use the same parameters
Analog line circuit packs use these parameters, according to the location parameters of the
circuit pack.
Companding for DCP telephones and circuit packs per location
You can administer the Companding Mode for each remote office, media gateway, and the rest
of the system that is circuit switched.
l
l
When a Digital Communications Protocol (DCP) telephone comes into service,
Communication Manager downloads the correct companding mode for the location of the
telephone.
When a circuit pack comes into service, Communication Manager downloads the
administered companding mode for the Avaya 8XXX Server, remote office, or media
gateway that is supporting that circuit pack.
Location ID in Call Detail Record records
You can administer the following CDR parameters in the custom CDR format for both the source
and destination:
l
location
l
time zone
l
country
Issue 6 May 2009
99
Localization
Loss plans per location
For each location, you can administer Digital Loss & Tone Loss, DCP terminal loss parameters,
and administrator-entered customizations.
When inserting loss for a multilocation intrasystem call, Communication Manager treats the call
as if IP tie trunks are connecting the different parties. When an audio stream is converted from
time-division multiplexing (TDM) to Internet Protocol (IP), the system adjusts the audio stream.
The system adjusts the audio stream by the IP media processor board of the sending location,
to an ISO standard level for voice over IP. The system then adjusts the audio stream by the
media processor of the receiving location to match the TDM levels for that location.
This board level adjustment is not done for DS1 remoted expansion port networks (EPNs). Use
DS1 remoted EPNs between countries only if the countries have similar voice transmit levels.
Multifrequency signaling per trunk group
Prior to Communication Manager release 2.1, you administer R2-Multifrequency Coded (MFC)
signaling parameters per system. With Communication Manager release 2.1 and higher, you
administer R2-MFC signaling parameters per trunk group.
R2-Multifrequency Coded signaling trunk groups use one of 8 sets of MFC signaling parameters
according to the MFC signaling code administered for that trunk group.
Tone generation per location
You can administer tone generation characteristics and administrator-entered customizations
per location. You can administer the server so that, when a telephone or trunk needs to play a
Communication Manager ECS-generated tone, the software plays a tone into the call using the
tone characteristics of the location of the listening endpoint, or of another endpoint on the call.
Public network call priority
Provides call retention, forced disconnect, intrusion, mode-of-release control, and re-ring to
switches on public networks. Different countries frequently refer to these capabilities by different
names.
100 Avaya AuraTM Communication Manager Overview
World class tone detection
World class tone detection
World class tone detection enables Communication Manager to identify and handle different
types of call progress tones, depending on the system administration. You can use the tone
detector and identification to display on data terminal dialing and to decide when to send digits
on trunk calls through abbreviated dialing, ARS, AAR, and data terminal dialing.
XOIP Tone Detection Bypass
The X over IP Tone Detection Bypass feature (where X = modem, fax, TTY-TDD, etc.) serves
customers using older or non-standard external equipment such as modems, fax, TTY devices
which are not easily recognized by VoIP resources within Communication Manager. By
identifying this external equipment through administration, VoIP firmware determines whether to
immediately attempt to put a call in pass-through mode, or allow the system to handle it
normally.
Issue 6 May 2009
101
Localization
102 Avaya AuraTM Communication Manager Overview
Chapter 9: Message integration
Avaya Aura™ Communication Manager Messaging
The CM Messaging application enhances communications and information exchange within
enterprises, helping customers be more successful with call answering and messaging. The CM
Messaging application enables customers to see messages on their PCs, add a voice mail
component to an e-mail, and listen to e-mail using voice mail. The distributed architecture in CM
Messaging is designed for reliability and survivability and is centrally managed for simplicity,
efficiency and quick response to help ensure business recovery.
The Communication Manager license includes the CM Messaging application with up to 500
mailboxes. CM Messaging enforces licensing for mailboxes in excess of 500.
CM Messaging supports digital (TCP/IP) networking protocol. More extensive networking can
be provided with the Avaya Interchange.
All the administrative features for CM Messaging are available on the Messaging Administration
Web interface. Using the Web interface, the administrator can perform a system backup and
restore of all administered data—announcements, recorded names, greetings—and
approximately 50 hours of messages over the local area network (LAN). The screens are easier
to understand and more intuitive, which reduces installation time and lessens the need for
training and experience. The CM Messaging system uses smart defaults rather than requiring
every field to be addressed.
Enhanced security features, such as disallowing voice PINS related to mailbox number, aging of
voice mail PINS, locking of voice mails after invalid PIN attempts, duration of voice mail locking,
default voice mail PIN length and so on.
Upgrade CM Messaging from release 3.x, 4.x, 5.x to release 5.2.
For more information on the CM Messaging messaging application, see the Avaya Aura™
Communication Manager Messaging Application Documentation CD collection on the Avaya
Support web site.
CM Messaging application
While many voice messaging systems require separate equipment and connections, the CM
Messaging application installs on the Communication Manager application to support advanced
voice messaging capabilities without the need for an adjunct processor. The messaging
capability on servers that support CM Messaging system are:
l
Support for S8500C Server with up to 5,000 mailboxes
l
Support for S8510 Server with up to 6,000 mailboxes
Issue 6 May 2009
103
Message integration
l
Support for S8300 Server with up to 450 mailboxes
l
Support for S8400 Server with up to 900 mailboxes
Whenever you call the CM Messaging system, you interact with it by entering commands
through your telephone touch-tone keypad. You simply specify the desired activity, and follow
the voice prompts for the desired task.
Special voice-processing features include voice mail, call answering, outcalling, multi-level
automated attendant, and bulletin board. The following is a summary of CM Messaging
capabilities:
l
l
l
l
l
l
l
l
l
l
l
l
Enhanced List application provides the ability to deliver messages to large numbers of
recipients.
Lockout Notification Mailbox is a mailbox that you can create to send out locked login
notification to a local mailbox.
Calling Party Number, when turned on in the Messaging Administration Web interface,
gets you up to 39 digits of the message sender's number when you listen to the message
header from Telephone User Interface, or when you use a IMAP4/POP3 client to retrieve
the call answer message. However, you need to administer Communication Manager to
get up to 39 digits of the calling party.
Support for * in the dial string.
Outcall scheduling for a subscriber is based on a subscribers time zone and not on the
system time zone.
Depending on the Privacy Enhancement Type set, the messaging system blocks a private
message and a user sees a warning or alert message when retrieving a private message
from a IMAP4/POP3 client. The language of alerting message depends on Message
Locale defined for that user.
Enhanced no cover 0 does not follow the coverage path for the covering extension.
However, you can press (*T) to transfer the call to another extension.
Messaging Capacity calculator helps to determine the number of call answer ports
required to administer the system.
Configure an external SMTP relay host using an IP host name to deliver an email that is
not addressed for delivery to a locally known CM Messaging system or an Intuity Audix LX
system that supports Internet Messaging. The external host is responsible to deliver the
email to its correct destination.
Configure a server with CM Messaging application installed on it as a Centralized
Messaging Server (CMS).
Shared extensions provide personal mailboxes for each person sharing a telephone.
Multiple personal greetings allows you to prepare a pool of up to nine personal greetings to
save time and provide more personal customer service. Separate messages can indicate
that you are on the telephone, away from the desk, on vacation, etc. You can assign
different messages to internal, external, or after-hours calls.
104 Avaya AuraTM Communication Manager Overview
Avaya Aura™ Communication Manager Messaging
l
l
l
l
l
l
l
l
l
l
l
l
l
l
l
l
l
l
l
Priority messaging places important messages ahead of others. Internal and external
callers can mark the message as priority.
Outcalling automatically dials a prearranged telephone number or pager when you have
messages in your voice mailbox.
Priority outcalling automatically dials a prearranged telephone number or pager when you
have priority messages in your voice mailbox.
Broadcasting allows you to send a single message to multiple recipients or to all users on
the system.
System broadcast allows you to send broadcast messages as regular voice messages, or
as messages that recipients hear as they log in.
CM Messaging directory allows you to look up the extension number of any other user by
entering their name on the telephone keypad.
Personal directory allows you to create a list of nicknames for quick access to telephone
numbers.
Call answering for nonresident subscribers provides voice mailboxes for users who do not
have an extension number on the system.
Full mailbox answer mode informs callers whenever messages cannot be left because
there is no room in a subscriber mailbox.
Name record by subscriber lets you record your own name on the system.
Automatic message scan can play all new messages in part or in their entirety without
requiring you to press additional buttons, which is particularly useful when you are getting
messages from your mobile telephone.
Sending restrictions by community enables you to limit the communities of callers who can
communicate using CM Messaging voice messaging.
Group lists allows you to create mailing lists of up to 250 people to use for broadcasting
messages.
Message forwarding allows you to forward messages with or without attached comments.
Name addressing allows you to address messages by name if you do not know the
extension.
Private messaging is a special coding feature that prevents recipients from forwarding
messages.
Leave word calling allows you to press a button on your telephone in order to leave a
standard “call me” message on any extension.
Multiple language support allows you to install up to nine languages on the system, from a
superset of 30 available languages.
Enhanced message handling gives you the flexibility for handling messages. Two of these
features are optional advance/rewind that lets you advance through and rewind individual
Issue 6 May 2009
105
Message integration
messages, and undelete messages that lets you retrieve any messages that you may
have accidentally deleted.
l
l
l
l
l
l
l
l
l
l
Fax messaging allows you to handle faxes as easily as you handle voice mail. You can
send, receive, store, scan, delete, skip, or forward faxes. This feature is fully integrated
with voice messaging, so you can attach faxes to voice messages, for example. You can
also create special mailboxes for each of your fax machines. These mailboxes accept fax
telephone calls when the fax machine is busy and then deliver the fax to the fax machine
when the fax machine is available.
Turn off CM Messaging call answering allows you to turn off call answering in order to
conserve system resources. You can create a message that tells callers they cannot leave
a message, giving them another number to call, for example.
Pre-addressing allows you to address a message before recording it.
Integrated messaging allows you access and manage incoming voice, fax, and e-mail
messages and file attachments from your personal computer or your telephone. A voice
message appears in your e-mail mailbox, for example, and vice versa. You can also set
options to have just the message headers appear in the alternate mailbox. You can also
create a voice or fax message by telephone and send it to an e-mail recipient.
Text-to-speech allows you listen to a voice rendering of text messages sent from a
supported e-mail system and/or message manager.
Print text allows you to print messages sent from a supported e-mail system and/or
message manager.
Enhanced addressing allows you to send a message to up to 1500 recipients.
Transfer restrictions allow you to control toll fraud by restricting transfers going through the
voice messaging system.
Internet messaging allows you to exchange messages (voice and text) with any e-mail
address via the World Wide Web.
International availability.
Record on messaging
Users can record conversations by pressing a single button using the Telephone User Interface.
This feature uses AUDIX as the recording device.
Note:
Note:
It it important that anyone who wants to activate this feature should study and
understand your local laws regarding the recording of calls before activating this
feature.
A feature button named audix-rec is used for this feature. The button is available for all
stations that have administrable feature buttons. When administered, the button also requires a
hunt group extension number (for the CM Messaging extension number) along with it.
106 Avaya AuraTM Communication Manager Overview
Audible message waiting
Note:
Note:
Attendant consoles do not have this button.
To record a conversation when a call is in process, press the audix-rec button. When you
push the button, the LED light for the feature button begins to flash. After about 4~6 seconds,
internal users who are participating in the call will notice that the telephone display changes to
CONFERENCE. The LED light on the telephone that initiated the recording is steadily illuminated.
This indicates that the AUDIX recording facility is ready and begins to record the conversation.
The internal users on the same switch with the display equipment can notice that the number of
parties in the call increases by 1. At this point, depending on the administration, a ready
indication tone will play to all the parties in the call, the initiator only, or none of the parties.
After enough information has been recorded, the initiator can then stop the recording by
pressing the audix-rec button a second time when the LED light is illuminated. The feature
button LED light on the initiator telephone goes out. The internal users with the display
equipment can again notice that the number of parties in the call decreases by 1. The call
remains active.
The Interval For Applying Periodic Alerting Tone field is used to allow the switch
administrator to choose an interval to play an alerting tone to all the parties on the call during
recording. Values are 0 to 60, and the default is 15. This means, if the default value is used, that
all parties on the call hear an alerting tone every 15 seconds that indicates the conversation is
being recorded. If the value for the field is 0, then no periodic tone is played during recording.
Audible message waiting
Audible message waiting places a stutter at the beginning of the dial tone when a telephone
user picks up the telephone. The stutter dial tone indicates that the user has a message waiting.
This feature is particularly useful for visually impaired people who may not be able to see a
message light. It is often used with telephones that have no message waiting lights.
Audible message waiting may not be available in countries that restrict the characteristics of dial
tones provided to users.
Leave Word Calling
Leave Word Calling (LWC) allows internal system users to leave a short preprogammed
message for other internal users. The preprogammed message usually is the word “call,” the
caller name, extension, and the time of the call. When the message is stored, the message
lamp on the called telephone automatically lights.
Issue 6 May 2009
107
Message integration
LWC messages can be retrieved using a telephone display, voice message retrieval, or
telephone user interface. Messages may be retrieved in English, French, Italian, Spanish, or a
user-defined language.
Leave Word Calling - QSIG/DCS
The Leave Word Calling (LWC) feature is extended to enterprise networks with QSIG as the
private network protocol, as well as those with DCS.
For enterprise networks that are mixed or in transition from DCS to QSIG, interworking of the
LWC feature between the protocols can be provided. LWC also works within a single
non-networked switch.
Note:
Note:
A DCS+ signaling group is needed, but can only be used in networks with 4-digit
or 5-digit dial plans.
Manual message waiting
This feature allows multi-appearance telephone users to light the status lamp associated with
the manual message waiting button at another multi-appearance telephone. They do this by
simply pressing a button on their own telephone. This feature can be administered only to pairs
of telephones such as a secretary and an executive. The secretary might press the button to
signal to the executive that a call needs answering or someone has arrived for an appointment.
The executive might use the button to indicate that he or she should not be disturbed.
Centralized voice mail (Tenovis)
A Tenovis C3000 Voice Mail server that was connected through a Tenovis system to a Avaya
AuraTM Communication Manager server, could not light the Message Waiting Indicator lamps
when messages were left for Avaya AuraTM Communication Manager users. Now, when
connecting a Centralized Voice Mail to the I55, it is possible to switch the message indication for
Avaya AuraTM Communication Manager subscribers in a QSIG Corporate Network.
108 Avaya AuraTM Communication Manager Overview
Message retrieval
Message retrieval
With the message waiting lamp on their telephones, employees always know when they have
messages. Messages can be retrieved in a variety of ways. These message retrieval options
can be assigned to individual users.
You can also use Message Manager PC software to retrieve messages.
Message Sequence Tracer enhancements
In the past, it had been difficult to trace messages through the Message Sequence Tracer
(MST) tool pertaining to a particular socket because there was no tag in each message
distinguishing it from other sockets.
New message formats for outgoing and incoming data now include the socket number/identifier.
These new formats use new Type identifiers of 05 and 06. A pair of new formats 07 and 08 have
also been created for outgoing and incoming socket control messages on the PROCR
ip-interface.
By creating new format types for these new formats, the task of decoding these messages is
easier.
The following enhancement was made to the Message Sequence Tracer (MST):
l
Signaling messages between Avaya AuraTM Communication Manager and the TN799
CLAN can now be traced for better diagnostics during network outages.
- Add processor TN799 CLAN socket information to the MST trace in order to help
developers debug socket problems.
- Enhance MST to include the socket number in socket data.
- Add TN799 CLAN board ID to CLAN MST IP socket trace messages.
Octel integration
Avaya AuraTM Communication Manager integrates with the entire line of Octel messaging
systems including the Octel 200/300 message server, and the Octel 250/350 message server.
Issue 6 May 2009
109
Message integration
QSIG/DCS voice mail interworking
QSIG/DCS voice mail interworking is an enhancement to the QSIG feature. It integrates DCS
and QSIG centralized voice mail using the DCS+/QSIG gateway. Switches labeled DCS+/QSIG
integrate multi-vendor PBXs into a single voice messaging system. QSIG/DCS voice mail
interworking works on G3r, G3si, and G3csi. It provides network flexibility, DCS functionality
without a dedicated T1.
Multiple QSIG voice mail hunt groups
Avaya AuraTM Communication Manager provides for ten message center hunt groups to
support QSIG integrated messaging. This feature allows customers to spread users in a single
Avaya AuraTM Communication Manager system over multiple messaging systems. This allows
users to move among Avaya AuraTM Communication Manager systems while retaining their
same voice mailbox. Users do not lose voice messages.
This feature also enhances customer usability of Avaya messaging systems in the enterprise by
allowing not only for growth, but the ability to migrate end users on a single Avaya AuraTM
Communication Manager system.
Voice mail retrieval button
Avaya Communication Manager supports the voice mail retrieval feature as a fixed feature
button on the 2420 DCP and the 4602 telephone.
A field, “voice-mail Number: _______” appears on the Station screen for stations of type 2420
and 4602. The allowed values for this field are identical to the values allowed for an autodial
feature button number. The field is a fixed field allowing entry of up to 16 digits that are
auto-dialed to access the user's voice mail system.
l
If the number field is blank, the voice mail retrieval button is treated like the “Transfer to
Voice Mail” button.
If the number field is not blank, the voice mail retrieval button is treated like an autodial button.
Voice message retrieval
Voice message retrieval allows telephone users, remote access users, and attendants to
retrieve leave word calling and call coverage voice messages. You can use voice message
110 Avaya AuraTM Communication Manager Overview
Voice messaging and call coverage
retrieval to retrieve your own messages or messages for another user. However, you can only
retrieve messages for another user:
l
from a telephone or attendant console in the coverage path
l
from an administered system-wide message retriever
l
if you are a remote-access user and you know the extension and associated security code
The system restricts unauthorized users from retrieving messages.
Voice messaging and call coverage
Often a CM Messaging system is set up as the last point on a call-coverage path, as shown in
Figure 3: Typical call coverage options on page 112. A secretary or colleague who answers a
redirected call intended for you can also transfer the caller to your CM Messaging mailbox. The
caller may prefer to leave voice-mail for you if the message is personal, lengthy, or technical.
Many other options are available. For example, a caller can redirect a call from the CM
Messaging system to an attendant. Or the caller can transfer to another extension instead of
leaving a message. You can even have the CM Messaging automated attendant answer all
calls to the company and send calls to various extensions. In this case, callers are instructed to
enter keypad commands to direct the call.
Issue 6 May 2009
111
Message integration
Figure 3: Typical call coverage options
Figure notes:
A
External call: active, busy, do not answer
1.
Secretary
B
Internal call: cover all
2.
Clerk
C
Internal call: active, busy, do not answer
3.
CM Messaging voice
messaging
D
Internal call: send all calls
4.
Message center group
112 Avaya AuraTM Communication Manager Overview
Chapter 10: Mobility
IP telephones or IP Softphones allow you to access the features of Communication Manager
without having to be tied to one location. One of the major benefits of IP telephones is that you
can move the telephones around on a LAN just by unplugging them and plugging them in
somewhere else. One of the main benefits of IP softphones is that you can load them on a
laptop PC, and then use the PC's modem to connect them to the switch from almost anywhere.
For more information, see Avaya IP Softphone on page 83.
Administration Without Hardware
This feature allows you to administer telephones that are not yet physically present on the
system. This feature works the same as administration with hardware: when stations are
moved, user-activated features such as call forwarding and send all calls are preserved and
functional. This greatly facilitates the speed of setting up and making changes to the telephones
on the system.
Automatic Customer Telephone Rearrangement
Automatic Customer Telephone Rearrangement (ACTR) allows a telephone to be unplugged
from one location and moved to a different location without additional switch administration. The
switch automatically associates the extension to the new port.
ACTR works with the 2420 DCP telephone, the 6400 serialized telephones, and newer DCP
telephones. The 6400 serialized telephone is stamped with the word “serialized” on the
faceplate for easy identification. The 6400 serialized telephone memory electronically stores its
own part ID (comcode) and serial number. ACTR uses the stored information and associates
the telephone with new port when the telephone is moved.
ACTR makes it easy to identify and move telephones.
Issue 6 May 2009
113
Mobility
Avaya Wireless Telephone Solutions
Avaya Wireless Telephone Solutions (AWTS) is fully integrated with Communication Manager,
and allows a user full access to Communication Manager features from a mobile telephone.
Note:
Note:
Avaya Wireless Telephone Solutions (AWTS) replaces the DEFINITY Wireless
Business System (DWBS).
Avaya Extension to Cellular
The Avaya Extension to Cellular feature provides the expansion of mobile services, including
one-number availability, increased user capacities, flexibility across facilities and hardware,
more control over unauthorized usage, enhanced enable/disable capability, increased
serviceability, and support of IP trunk facilities.
Avaya Extension to Cellular and off-PBX stations (OPS) provides users with the capability to
have one administered telephone that supports Communication Manager features for both an
office telephone and one outside telephone. Extension to cellular/OPS allows users to receive
and place office calls anywhere, any time. People calling into an office telephone can reach
users even if they are not in the office. Users could receive the call on their cell telephone, for
example.
This added flexibility also allows them to use certain Communication Manager features from a
telephone that is outside the telephone network.
Previous versions of Extension to Cellular allowed for office calls to be extended to the cell
telephone of a user. Also, calls from the cell telephone would appear as if the call originated
from the user office telephone when calling another telephone on the same call server. Certain
features within Communication Manager are available from the cell telephone. These features
are still available.
In previous versions of Extension to Cellular, cell telephones had to be administered as
XMOBILE stations. This is no longer necessary with Communication Manager Release 2.0.
If you had administered Extension to Cellular in earlier releases of Communication Manager,
you do not have to change the administration to continue using Extension to Cellular features. It
still works. However, users would not have the full range of features that are now possible with
Extension to Cellular/OPS.
You can gain multi-country switch access by administering multiple sets of Feature Name
Extension (FNE). You can have one Communication Manager system with different gateways in
multiple locations. Each gateway can have its own FNE set.
114 Avaya AuraTM Communication Manager Overview
Avaya Extension to Cellular
In addition, Communication Manager supports the ability for Public Fixed Mobile Convergence
(PBFMC) calls appear to be on the same location as a user’s desk set and not the gateway that
supports incoming trunks. The same location based dialing rules can be used to ensure that the
same ARS and route patterns are used for calls in the same location.
Communication Manager also supports the use of the Confirmed Answer option for cellular
voice mail avoidance for any OPTIM application, including EC500. With Confirmed Answer,
upon answering the phone, the user hears the dial tone.The user must then press one of the
digits on the cellular phone's keypad. Until the system receives a digit, the system does not treat
the call as answered.
There are several uses for Confirmed Answer:
l
l
In some businesses with the use of EC500 (such as for after hours support), it is critical
that a call be treated as answered only if a person answers the call. In such a scenario,
Confirmed Answer is the only reliable voice mail avoidance method.
Since Confirmed Answer is the most reliable form of cellular voice mail avoidance, some
users may be willing to use the feature with PBFMC. An added benefit of the feature is that
the dial-tone is a signal to the user that this is a business call, not a personal call.
Communication Manager provides the following Extension to Cellular features:
l
l
l
l
l
Conditional Call Extending (controls which type of calls to extend when EC 500 is enabled)
Shared Voice Connections (allows two voice calls to share a single trunk connection
between the cell phone and the PBX)
Sharing Mappings among CM PBXs (allows the station name and station mapping
information to be across multiple Communication Managers)
SPFMC OPTIM Application (enables support for dual-mode cell phones)
One-X Mobile Server and Application Support (allows use of one-X server to configure and
control a set of features; each extension can support four one-X applications)
Off-PBX station
The off-PBX station (OPS) application type is used to administer of a SIP telephone. OPS
cannot be disabled using the Extension to Cellular enable/disable feature button.
Note:
Note:
A 4602 SIP telephone must register with the SIP proxy regardless of whether
OPS is administered.
The Extension to Cellular/OPS application allows for many of the parameters used for the
original Extension to Cellular application to be ported onto one of several DCP and IP station
types. From a call processing perspective, Extension to Cellular/OPS is in fact dealing with a
multi-function telephone, whereas the previous Extension to Cellular implementation utilized
one or two XMOBILE stations that behaved like analog station types.
Issue 6 May 2009
115
Mobility
The Extension to Cellular/OPS application has its own documentation set. For a complete list of
Extension to Cellular/OPS documentation, see your Avaya representative.
E911 ELIN for IP wired extensions
This feature automates the process of assigning an emergency location information number
(ELIN) through an IP subnetwork (“subnet”) during a 911 call. The ELIN is then sent over either
CAMA or ISDN PRI trunks to the emergency services network when 911 is dialed. This feature
properly identifies locations of wired IP telephones that call an emergency number from
anywhere on a campus or location.
Note:
This feature depends upon the customer having subnets that correspond to
geographical areas.
Note:
This feature works for both types of IP endpoints:
l
H.323
l
SIP
A caller who needs emergency assistance dials a Universal Emergency Number - for example,
911 in the United States, 000 in Australia, and 112 in the European community. The call routes
through a local Central Office, through an emergency tandem office, to the appropriate Public
Safety Answer Point (PSAP). The PSAP answers the call.
A typical tandem office can route the call to a PSAP within at most four surrounding areas. (In
the US, that translates to four surrounding area codes.) If the PSAP that receives the call is not
the correct one to handle the emergency, the PSAP might be able to transfer the call to the
correct PSAP. Such transfers can only occur between geographically adjacent or nearby
PSAPs.
Each PSAP usually covers one city or one rural county. At the PSAP, emergency operators
determine the nature of the emergency and contact the appropriate agency: police, fire,
ambulance, etc. A single PSAP is usually responsible for an area covering several independent
police and fire departments in the United States.
With Enhanced 911 (E911), the system might send to the emergency services network the
Calling Party Number (CPN) with the call over Centralized Automatic Message Accounting
(CAMA) trunks or through the Calling Number IE over ISDN trunks. A system at the PSAP uses
the CPN to look up the documented street address location of the caller from the Automatic
Location Information (ALI) database. The ALI database is usually owned and managed by Local
Exchange Carriers. Many enterprise customers choose to contract with a third party to update
the ALI database for them.
This depends on the assumption that a CPN always corresponds to the street address that the
system owner arranged to have administered into the ALI database. This assumption is not
always true.
116 Avaya AuraTM Communication Manager Overview
Personal Station Access
l
l
Users who have H.323 IP telephones can move them without notifying the system
administrator.
Users who have SIP IP telephones can use the same extension number simultaneously at
several different telephones.
Without this feature, if these users dial 911, the emergency response personnel might go to the
wrong physical location. With this feature, the emergency response personnel can now go to
the correct physical location. In addition, emergency response personnel can now go to the
correct physical location if a 911 emergency call comes from a bridged call appearance.
E911 device location for IP telephones
Communication Manager works with an E911 Manager device from RedSky Technologies. This
third-party E911 Manager provides a flexible, complete, and automated E911 management
system for customers who want to implement voice over IP (VoIP) telephony. The E911
Manager from RedSky Technologies works with Communication Manager release 2.1 and
beyond to keep the Automatic Location Information (ALI) record for each extension correct. The
E911 Manager also provides notification whenever someone moves an IP endpoint to a new
subnet.
Personal Station Access
The Personal Station Access (PSA) feature allows you to transfer your telephone station
preferences and permissions to any other compatible telephone. This includes the definition of
telephone buttons, abbreviated dial lists, and class of service, and class of restrictions
permissions.
PSA has several telecommuting applications. For example, several telecommuting employees
can share the same office on different days of the week. The employees can easily and
remotely make the shared telephone “theirs” for the day.
PSA association and dissociation are enhanced by Hot Desking. For more information, see Hot
Desking Enhancement on page 209.
Do not answer reason code
The Personal Station Access (PSA) feature uses Administration Without Hardware (AWOH), a
feature that allows the system administrator to assign a telephone without specifying a physical
port. For example, use “X” as the port. If a telephone is disassociated, it means that it is not
currently mapped to a particular physical telephone, such as a digital telephone. If a caller dials
Issue 6 May 2009
117
Mobility
into an extension that is currently disassociated, they are provided a message that indicates
“Don't answer” instead of “Busy”.
Name/number permanent display
When a person uses PSA to associate their extension with a station, a display appears on the
station indicating their name and extension number. This information is displayed until the user
disassociates their extension from the station using the PSA-disassociate feature access code.
Seamless Converged Communication Across Networks
The SCCAN (Seamless Converged Communication Across Networks) program consists of the
joint collaboration of three companies, Motorola, Proxim, and Avaya. The purpose of SCCAN is
to create and deploy a converged cellular Wireless Local Area Network (WLAN), and Internet
Protocol (IP) Telephony services that delivers new levels of communications mobility and
network connectivity.
The SCCAN system consists of new products including:
l
l
A Wi-Fi/cellular dual-system telephone from Motorola (sometimes referred to as the
Subscriber Unit)
Session Initiation Protocol (SIP)-enabled IP Telephony software from Avaya (most often
referred to as Communication Manager SIP Trunking)
l
IPSec security client from Avaya (VPNRemote) that runs on the Motorola handset
l
Light Access Points (LAPs) that are developed by Proxim
SIP Visiting User
The SIP Visiting User (SIP VU) feature enables users with the 9620 or 9630 SIP telephone to
log in to any SIP telephone in the enterprise and receive their own individualized services,
including menus, contacts, and buddy lists. When a SIP Visiting User makes an emergency call,
the call is routed to the local emergency personnel closest to the visited site, not the user’s
home site.
The SIP Visiting User feature relies on specialized firmware on the telephone, and also requires
SIP VU administration.
SIP Visiting Users can be roaming or non-roaming.
118 Avaya AuraTM Communication Manager Overview
Terminal Translation Initialization
l
l
Non- roaming visiting user - If a visiting user is non-roaming, the user is logged into a
visiting phone that is served by the user’s usual SIP Enablement Services (SES) home
server. For example, a non-roaming visiting user logs into a phone that is in the office
adjacent to their primary phone. No special connections need to be made to serve up the
contacts, permissions, and buddy list the non-roaming visiting user expects.
Roaming visiting user - If a visiting user is roaming, the visiting user is logged into a phone
that uses a SES home server that is different from the visiting user's home SES.
During registration, the roamed-to SES home server retrieves the user’s credentials from
the SES data service as the means to enable roaming for that user. This causes the user's
home SES to flag the user as roaming.
Terminal Translation Initialization
Communication Manager provides Terminal Translation Initialization (TTI), a feature that works
with Administration Without Hardware (AWOH). TTI associates the terminal translation data
with a specific port location through the entry of a special feature-access code, a TTI security
code, and an extension number from a terminal that is connected to a wired (but untranslated)
jack.
TransTalk 9000 digital wireless system
The TransTalk 9000 is a single-zone or dual-zone, in-building wireless system that provides a
mobility solution on Communication Manager-based systems. It delivers the benefits and
accessibility of a wireless telephone with all the power and functionality of a wired desk
telephone.
X-station mobility
X-station mobility allows remote users to access switch features. That is, X-station mobility
allows certain OEM wireless telephones remoted over a PRI trunk interface to be controlled by
Communication Manager as if the telephones were directly connected to the switch.
The telephones are administered to be of the type XMOBILE and have additional administration
information on the Station screen that assigns the capabilities of a remote station to the
associated PRI trunk group. The wireless telephones thus have access to such features as
call-associated display, bridging, message waiting, call redirection, and so forth.
Issue 6 May 2009
119
Mobility
X-station mobility is currently used for non-cellular wireless offers (DECT and PHS) in EMEA
and APAC regions, and the Extension to Cellular offer globally.
120 Avaya AuraTM Communication Manager Overview
Chapter 11: Port network and gateway connectivity
Asynchronous Transfer Mode
The Asynchronous Transfer Mode (ATM) switch is a replacement option for the CSS or the
direct-connect switch. Several Avaya ATM switch types can provide Communication Manager
port network connectivity. Non-Avaya ATM switches that comply with the ATM standards set by
the European Union can also provide Communication Manager port network connectivity.
Port Network Connectivity
ATM Port Network Connectivity (ATM-PNC) provides an alternative to the Center Stage Switch
(CSS) configurations for connecting the Processor Port Network (PPN) to one or more
Expansion Port Networks (EPN). ATM-PNC replaces the CSS in a DEFINITY R8r and later
network with an ATM switch or network. ATM-PNC is available with all three Communication
Manager reliability options - standard, high, and critical. In addition, it offers ATM-PNC
duplication.
ATM-PNC integrates delivery of voice, video, and data via ATM over a converged large
bandwidth network, providing reduced infrastructure cost and improved network manageability.
ATM-PNC uses standards-based open interfaces that can be provisioned with either new or
existing systems running Communication Manager.
Port Network Connectivity over WAN
ATM-PNC over a public Wide Area Network (WAN) represents an environment where the
customer uses a service provider's ATM network between privately-owned ATM switches. The
customer does not control the ATM switches in the network, including traffic policing policies
and product quality.
Using a public WAN, Permanent Virtual Paths (PVP) may be set up between customer-owned
ATM switches similar to the dedicated circuits in a private WAN. However, ATM cell processing
occurs in a public WAN so the customer is dependent on ATM switches owned and managed by
the service provider.
Switched Virtual Circuits (SVC) use the ATM protocol to transmit voice-like applications over
ATM networks. The advantage of the SVC solution is that Communication Manager can
dynamically signal the ATM network to provide more bandwidth as needed to handle peaks in
the call traffic. If the ATM network cannot handle the additional traffic, calls will be denied.
Issue 6 May 2009
121
Port network and gateway connectivity
Circuit switched
Center Stage Switch
Communication Manager supports CSS as a method to interface between the PPN and EPNs
using circuit switched technology to carry the voice traffic.
Center Stage Switch separation
Avaya 8XXX Servers in an Avaya MCC1, SCC1, or G650 Media Gateway configuration, with
four or more Port Networks (PN), use a Center Stage Switch (CSS) to interconnect the PNs.
The Center Stage Switch (CSS) separation feature allows for the physical separation of
redundant Avaya 8XXX Servers, and their corresponding CSS, to improve their survivability.
Avaya 8XXX Servers and the CSS can be separated up to 6.2 miles (10 km), providing backup
and survivability for a communications network in one or more remote locations.
Internet Protocol
H.248 media gateway control
Communication Manager uses standards based H.248 to perform call control to Avaya media
gateways such as the G700. H.248 defines a framework of call control signaling between the
intelligent Avaya 8XXX Servers and multiple “unintelligent” media gateways.
122 Avaya AuraTM Communication Manager Overview
Internet Protocol
Inter-Gateway Alternate Routing
For single-server systems that use the IP-WAN to connect bearer between port networks or
media gateways, Inter-Gateway Alternate Routing (IGAR) provides a means of alternately using
the public switched telephone network (PSTN) when the IP-WAN is incapable of carrying the
bearer connection. IGAR may request that bearer connections be provided by the PSTN under
the following conditions:
l
The number of calls allocated or bandwidth allocated via Call Admission
Control-Bandwidth Limits (CAC-BL) has been reached
l
VoIP RTP resource exhaustion in a MG/PN is encountered
l
A codec set is not specified between a network region pair
l
Forced redirection between a pair of network regions is configured
IGAR takes advantage of existing public and private network facilities provisioned in a network
region. Most trunks in use today can be used for IGAR. Examples of the better trunk facilities for
use by IGAR are:
l
Public or Private ISDN PRI/BRI
l
R2MFC
IGAR provides enhanced Quality of Service (QoS) to large distributed single-server
configurations.
Network Region Wizard
For large distributed single-server systems that have multiple network regions, the Network
Region Wizard (NRW) simplifies and expedites the provisioning of multiple IP network regions,
including Call Admission Control using Bandwidth Limits (CAC-BL) and Inter-Gateway Alternate
Routing (IGAR).
IP Port Network Connectivity
Communication Manager allows Control Channel Message Set (CCMS) messages to be
packetized over IP LAN and WAN connections to control multiple port networks.
Improved Port Network Recovery from Control Network Outages
The IP-connected port networks experience disproportionately long outages from short network
disruptions. The Improved Port Network Recovery feature provides customers using IP
connected Port Networks with less downtime in the face of IP network failures.
Issue 6 May 2009
123
Port network and gateway connectivity
The feature lessens the impact of network failures by:
l
l
Improving TCP recovery times that increase the IPSI-CM connection bounce coverage
time from the current 6-8 seconds range for the actual network outage to roughly 10
seconds. Results vary based on traffic rates.
Modifying the IPSI recovery action after a network outage to be a warm interrupt rather
than a IPSI application reset (hardware interrupt)). This prevents H.323 IP telephones from
having to re-register and/or have their connections regenerated. This minimizes recovery
time from network outages in the range of 15-60 seconds.
This feature also monitors the IPSI-CM connection and helps in identifying and troubleshooting
network related problems.
Link Recovery
IP calls must have an H.248 link between the Avaya G700 Media Gateway and the call
controller. The H.248 link between an Avaya server running Communication Manager and the
Avaya media gateway provides the signaling protocol for:
l
Call setup
l
Call control (user actions such as Hold, Conference, or Transfer)
l
Call tear-down
If the link fails for any reason, the Link Recovery feature preserves any existing calls and
attempts to re-establish the original link. If the gateway cannot reconnect to the original server,
then Link Recovery automatically attempts to connect with alternate TN799DP (CLAN) circuit
packs within the original server configuration or to a Local Spare Processor (LSP).
Link Recovery does not attempt to recover or overcome any network failure that created the link
outage. Link Recovery also does not diagnose or repair the network failure that caused the link
outage.
Since there is no communication possible between the Media Gateway and call controller
during a link outage, button depressions are not recognized, feature access does not work, and
neither does any other type of call handling. In essence, the system is unresponsive to any
stimuli until the H.248 link is restored. This might be the only indication that a Link Recovery is
in process.
! CAUTION:
CAUTION:
If an administrator attempts to add a telephone to a gateway while that gateway
is in Link Recovery, that station is not put into service when the gateway comes
back. If this happens, perform a busyout/release command on that telephone
when the gateway comes back into service.
124 Avaya AuraTM Communication Manager Overview
Separation of Bearer and Signaling
Feature highlights:
l
l
l
l
l
Call signaling channel failures are detected at a fast rate (in the order of 30 seconds, by
default).
The endpoint has an awareness about the primary and the alternate gatekeepers for the
purpose of faster and less disruptive recovery from signaling channel failures.
The endpoint attempts to re-establish the signaling channels with the primary gatekeeper
while preserving an existing call.
An IP endpoint's registration (while it is recovering from a signaling failure) can be
accepted while preserving that endpoint's existing call(s).
The customer can administer the endpoint recovery parameters (such as timers and
gatekeepers).
In order for IP endpoints to take advantage of this feature, the firmware or application software
must be updated with the new algorithm that supports the resiliency feature. IP endpoints
include IP telephones and IP softphones. However, since the feature provides backward
compatibility, it is not mandatory that existing IP endpoints be upgraded.
Separation of Bearer and Signaling
The Separation of Bearer and Signaling (SBS) feature provides a low cost virtual private
network with high voice quality for customers who cannot afford private leased lines. SBS
provides a DCS+ VPN replacement for those customers needing Dial Plan Expansion (DPE)
functionality.
Note:
DCS does not work with six-digit or seven-digit dial plans. Although QSIG does
work with six-digit and seven-digit dial plans, QSIG does not work over VPNs.
Note:
The SBS feature supports:
l
QSIG private networking signaling over a low cost IP network
l
Voice (bearer) calls over public switched network
l
Association between QSIG feature signaling information and each voice call
You must always use AAR/ARS/UDP to originate an SBS call. You cannot use a Trunk Access
Code / Dial Access Code to originate an SBS call.
Proper administration and configuration is required for SBS calls to work correctly. This
includes:
l
Fields in the System-Parameters Features screen, a field on the Trunk Group screen,
and a Station type called an SBS Extension (an extension number without hardware
assigned to it that is used to associate the separate bearer and signaling calls).
Issue 6 May 2009
125
Port network and gateway connectivity
l
l
Customers must allocate a sufficient number of SBS extensions based on expected SBS
traffic volume. The same applies to SBS trunk group members.
Each administered SBS extension must correspond to a DID/DDI number obtained from a
local service provider.
Note:
Obtaining a DID/DDI number for each SBS extension is not necessary if the
Feature Plus Pseudo DID feature is available.
Note:
l
l
In remote office configurations or other remote gateway configurations with limited direct
network access, these DID/DDI numbers should be obtained from a service provider that
is local to the controlling gateway server, not local to the remote office/gateway. This
eliminates excessive traffic through the remote office/gateway to its controlling gateway
server.
The ISDN Public-Unknown Numbering screen must be correctly administered to map
every SBS extension to the corresponding national public network complete number (that
is, the DID/DDI number). This screen is used to develop the complete number even if the
incoming SBS trunk group numbering format is administered for private numbering.
126 Avaya AuraTM Communication Manager Overview
Chapter 12: Trunk connectivity
Asynchronous Transfer Mode
See Asynchronous Transfer Mode on page 121.
Circuit Emulation Service
ATM-circuit emulation service (ATM-CES) lets Communication Manager emulate ISDN-PRI
trunks on an ATM facility. These virtual trunks can serve as integrated access, tandem, or tie
trunks.
ATM-CES trunk emulation maximizes port network capacities by consolidating trunking. For
example, the CES interface can define up to eight virtual circuits for tie-line connectivity,
consolidating onto one circuit card network connectivity that usually requires multiple circuit
packs.
CMS measurement of ATM
See CMS measurement of ATM on page 63.
Circuit switched
DS1 trunk service
Bit-oriented signaling that multiplexes 24 channels into a single 1.544-Mbps stream. DS1 can
be used for voice or voice-grade data and for data-transmission protocols. E1 trunk service is
bit-oriented signaling that multiplexes 32 channels into a single 2.048-Mbps stream. Both DS1
and E1 provide a digital interface for trunk groups. Digital Service 1 (DS1) trunks can be used to
provide T1 or ISDN Primary Rate Interface (PRI) service.
Issue 6 May 2009
127
Trunk connectivity
Echo cancellation - with UDS1 circuit pack
The universal DS-1 (UDS1) circuit pack (TN464GP/TN2464BP) available for all Communication
Manager platforms has echo cancellation circuitry. The echo cancellation capability of the circuit
pack is intended only for channels supporting voice communication. It is not desirable to provide
echo cancellation over channels supporting data communication.
The TN464GP/TN2464BP is intended for Communication Manager customers who are likely to
encounter echo over circuits connected to the public network. The occurrence of echo is likely if
Communication Manager is configured for complex services such as ATM or IP. In addition,
echo is likely to occur if Communication Manager interfaces to local service providers who do
not routinely install echo cancellation equipment in all their circuits.
E1
Communication Manager also supports E1 connections. T1/E1 access and conversion allows
simultaneous connection to both T1 (1.544 Mbps) and E1 (2.048 Mbps) facilities (using
separate circuit packs).
T1
When planning your networking requirements, one of the options you should consider is
multiplexing over digital services 1 (DS1) facilities.
Separate licensing for TDM stations and TDM trunks
Prior to release 2.0, Communication Manager was sold by licensed ports that included stations
and trunks. The system displayed the total of licensed ports in the Maximum Ports field on the
Optional Features screen.
For Communication Manager, Avaya sells licenses for stations, but not for trunks. Currently, the
Maximum Ports field on the Optional Features screen is used for licensing ports, which
include both trunks and stations.
In Communication Manager, a separate field, Maximum Stations, is created on the Optional
Features screen to track station licenses only. This helps customers easily identify the number
of station licenses on the system.
128 Avaya AuraTM Communication Manager Overview
Internet Protocol
Internet Protocol
H.323 trunk
A TN802B in MedPro mode or a TN2302AP IP interface enables H.323 trunk service using IP
connectivity between two systems running Communication Manager. The H.323 trunk groups
can be configured as system-specific tie trunks, generic tie trunks, or direct-inward-dial (DID)
public trunks. In addition, the H.323 trunks support ISDN features such as QSIG and BSR.
Improved button downloads
The improved button download to IP endpoints feature greatly reduces the time needed for
button labels to be sent to the IP telephone upon a new registration. This feature also reduces
the traffic being sent to the IP telephones when button labels have not changed on a
reregistration.
The Avaya H.323-based IP telephones request whether they need feature button information
when they send the RRQ message to the Communication Manager server during registrations
or re-registrations. If the IP telephones request the button information, or if Communication
Manager is aware of an administration change that occurred to change button information,
Communication Manager then downloads that button information (including the labels) in the
RCF message back to the telephone. This process does not rely on background maintenance
sending down the labels in CCMS messages, or on NSM messages after registration. If the IP
telephones did not request the button information (due to a re-registration), and no recent
administration changes in the button information had occurred, Communication Manager does
not include any button information in the RCF message. The IP telephones use the labels they
already have in memory.
Increased trunk capacity
Communication Manager trunk capacities, which include all IP and TDM trunk types, are
increased from 8,000 to 12,000 for any single Communication Manager server within a
configuration. With an Avaya S8730 or S8720 server in an XL configuration, Communication
Manager trunk capacities support 12,000 trunks, multiplied by the number of Communication
Manager servers. The maximum trunk capacity is 96,000 (12,000 trunks x 8 Communication
Manager servers). For information on trunk capacities, see Avaya Aura™ Communication
Manager System Capacities Table, 03-300511.
Issue 6 May 2009
129
Trunk connectivity
IP loss groups
A primary reason to accomplish a loss plan for voice communication systems is the desire to
have the received speech and tone loudness at a comfortable listening level. This should be
accomplished so that users can listen to each other without being concerned who or where the
remote party is, or what kind of telephone equipment each may be using.
A connection with an end-to-end loss (called an Overall Loudness Rating) of 10 dB - which
approximates a normal conversation between a talker and listener spaced one meter
apart - provides a high degree of satisfaction for the majority of users. Therefore, voice
communication standards for end-to-end loss are based on this number.
Communication Manager has now defined two additional loss groups for IP telephony. The
purpose of these two loss groups is to set speech and tone loudness separately for IP
connections. These loss groups use country-specific gateway loss plans.
The two IP loss groups are:
l
Loss Group 18: IPtrunk - loss group for IP trunks (IP Carrier Medium)
l
Loss group 19: IPphone - loss group for IP terminals (IP ports)
On an upgrade, if the default for an IP station loss plan is 2, and the IP trunk loss plan is 13,
Communication Manager changes the defaults to 19 and 18 respectively.
IP trunks
IP trunk groups may be defined as virtual private network tie lines between systems or ITS-E
servers running Communication Manager. Each IP trunk circuit pack provides a basic 12-port
package that can be expanded up to a total of 30 ports. The number of ports that are defined
will correspond to the total number of simultaneous calls transmitted over the IP trunk interface.
The benefits of IP trunk include a reduction in long distance voice and fax expenses, facilitating
global communications, providing a full function network with data and voice convergence and
optimizing networks by using the available network resources.
IP trunking is a good choice for basic, corporate voice and fax communications, where cost is a
major concern. IP trunk calls travel over a company intranet rather than the public telephone
network. So, for the most common types of internal corporate communications, IP trunks offer
considerable savings.
IP trunking is usually not a good choice for applications where calls have to be routed to multiple
destinations (as in most conferencing applications) or to a voice messaging system. IP trunk
calls are compressed to save network bandwidth. Repeated compression and decompression
results in a loss of data at each stage and degrades the final quality of the signal.
130 Avaya AuraTM Communication Manager Overview
Internet Protocol
The maximum number of compression cycles acceptable on a call is three, and three
compression cycles can compromise voice quality. Normal corporate voice or fax calls typically
go through fewer than three compression cycles. However, multipoint conference calls and
most voice messaging systems add too many compression cycles for acceptable quality.
IP trunk fallback to PSTN
The PSTN fallback of IP trunks feature refers to bypassing, or skipping over, IP trunks when
IP network conditions make the voice quality of IP trunks unacceptable. The skipping over of
IP trunks allows existing route patterns to look at secondary trunk groups for outgoing calls.
When these secondary trunk groups are not IP, the PSTN fallback feature applies. The key to
the existing implementation is the evaluation of the current network condition.
Evaluation data is obtained from ICMP PING packets that are generated from the TN2302/
TN2602 media processor circuit packs. Reports are generated from the circuit packs that
aggregate many individual ICMP PING requests.
The existing family of H.248 media gateways (G250, G350, and G700) do not provide this
functionality. In systems which do not contain TN media processors, or where those media
processors don't have access to the network regions that are reachable by H.248 gateways,
there is no PSTN fallback capability for IP trunks in the event of network degradation.
The PSTN fallback for IP trunks feature for H.248 media gateways resolves this problem.
IP trunk link bounce
H.323 trunk link bounce provides customers with fewer call failures in the event of an IP network
failure or disruption. This feature lessens the impact of IP network failures and disruptions by
postponing corrective action after an H.323 signaling link failure. When a failure is detected,
Communication Manager initiates a timer to allow time for the network to recover. If the network
does recover within that period, calls are allowed to continue without interruption.
Session Initiation Protocol
Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP
network.
SIP has a separate set of documentation. For more information on SIP, click the
documentation link from the http://www.avaya.com Web site.
Issue 6 May 2009
131
Trunk connectivity
SIP trunks
SIP trunking functionality allows a Linux server to function as a POTS gateway between
traditional legacy endpoints (stations and trunks) and SIP endpoints. It also provides SIP to SIP
routing. In the routing scenario, the server supports call routing similar to what a SIP proxy
would provide.
SIP links can be secured using TLS to encrypt signaling, and use Digest Authentication to
perform validation. When using TLS, the Media Encryption feature is also available to encrypt
audio channels.
SIP trunking functionality:
l
l
l
Provides access to less expensive local and long distance telephone services, plus other
hosted services from SIP service providers
Provides presence and availability information to members of the enterprise and
authorized consumers outside the enterprise, including other enterprises and service
providers
Facilitates SIP-enabled converged communications applications within the enterprise,
such as the Seamless Service Experience.
Allowing encryption of signaling and audio channel provides the customer with the option to
provide a secure communications infrastructure. See Signaling encryption for SIP trunks on
page 206.
Auxiliary trunks
Auxiliary trunks connect devices in auxiliary cabinets with Communication Manager. Some of
the features that are supported with this type of trunk are recorded announcements, telephone
dictation service, malicious call trace, and loudspeaker paging.
Advanced Private Line Termination
Provides access to and termination from CO (Central Office)-based private networks; namely,
Common Control Switching Arrangements (CCSA) and Enhanced Private Switched
Communications Service (EPSCS).
APLT trunks are physically the same as those used for analog tie trunks, where the trunk
signaling is compatible with EPSCS and CCSA network switches. The outgoing APLT trunk
repeats any number of digits to the private network as dialed. APLT trunks can tandem through
the PBX from EPSCS network only; CCSA networks require an Attendant to complete the call.
132 Avaya AuraTM Communication Manager Overview
Central Office
Central Office
Central Office (CO) trunks connect Communication Manager to the local central office for
incoming and outgoing calls.
Central Office support on G700 Media Gateway - Russia
See Central Office support on G700 Media Gateway on page 97.
Digital multiplexed interface
The digital multiplexed interface feature supports two signaling techniques: bit-oriented
signaling and message-oriented signaling for direct connection to host computers.
Digital multiplexed interface offers two major advantages:
l
l
digital multiplexed interface delivers a standard, single-port interface for linking host
computers internally and externally through a T1 carrier.
Since it is compatible with ISDN standards and is licensed to numerous equipment
manufacturers, digital multiplexed interface promotes multi-vendor connectivity.
Communication Manager supports two versions of digital multiplexed interface, each differing in
the way information is carried over the 24th channel:
l
Bit-oriented signaling
l
Message-oriented signaling
Bit-oriented signalling
Digital multiplexed interface bit-oriented signalling carries framing and alarm data and signalling
information for connections to host computers and other vendor equipment.
Message-oriented signalling
Digital multiplexed interface message-oriented signalling, fully compatible with ISDN-PRI, uses
the same message-oriented signalling format - link access procedure on the D-channel - as
ISDN-PRI for control and signalling. These signalling capabilities extend the advantages of
Issue 6 May 2009
133
Trunk connectivity
digital multiplexed interface message-oriented signalling multiplexed communications to the
public ISDN network.
Direct Inward Dialing
Direct Inward Dialing (DID) trunks connect Communication Manager to the local central office
for incoming calls dialed directly to stations without attendant assistance.
Direct Inward/Outward Dialing
Traditionally, Central Office (CO) trunks and Direct Inward Dialing (DID) trunks interface an
attendant console with a central office. A CO trunk services outgoing calls and accepts
incoming calls that are terminated at the attendant. A Direct Inward/Outward Dialing (DIOD)
trunk is used for calls that need to be terminated without an attendant interaction.
E&M signaling - continuous and pulsed
See E&M signaling - continuous and pulsed on page 98.
E911 CAMA trunk group
This screen administers the Centralized Automatic Message Accounting (CAMA) trunks and
provides Caller Emergency Service Identification (CESID) information to the local enhanced
911 system through the local central office.
Foreign Exchange
Foreign Exchange (FX) trunks connect Communication Manager to a Central Office other than
to the local office.
134 Avaya AuraTM Communication Manager Overview
ISDN trunks
ISDN trunks
Gives you access to a variety of public and private network services and facilities. The ISDN
standard consists of layers 1, 2, and 3 of the Open System Interconnect (OSI) model. Systems
running Communication Manager can be connected to an ISDN using standard frame formats:
Basic Rate Interface (BRI) and the Primary Rate Interface (PRI).
An ISDN provides end-to-end digital connectivity and uses a high-speed interface that provides
service-independent access to switched services. Through internationally accepted standard
interfaces, an ISDN provides circuit or packet-switched connectivity within a network and can
link to other ISDN supported interfaces to provide national and international digital connectivity.
Automatic Termination Endpoint Identifier
The user side will support automatic TEI assignment by the network. Both fixed and automatic
TEI assignment will be supported on the network side.
Call-by-call service selection
Enables a single ISDN-PRI trunk group to carry calls to a variety of services, rather than
requiring each trunk group to be dedicated to a specific service. It allows you to set up various
voice and data services and features for a particular call.
ETSI functionality
The full set of ETSI public-network and private-network ISDN features is officially supported.
This includes Look-Ahead Interflow (LAI), look-ahead routing, and usage allocation.
Also included is all QSIG supplementary services, such as:
l
Name identification
l
Call diversion (including rerouting)
l
Call transfer
l
Path replacement
ETSI functionality does not include:
l
DCS
l
Non-facility associated signaling
Issue 6 May 2009
135
Trunk connectivity
l
D-channel backup
l
Wideband signaling
ETSI completion of calls
The auto-callback feature known as ETSI Completion of Calls to Busy Subscriber (CCBS) and
on No Reply (CCNR) works over the DSS1 network. An analog phone actives this feature by
dialing the FAC. A multi-appearance voice terminal can use either the FAC or programmed
buttons. By pressing an idle Automatic Callback button, a caller can activate either the
SS-CCBS when the called station is busy, or the SS-CCNR when the called station does not
answer.
The originating caller receives a callback when:
l
l
The busy station becomes available after the user hangs up.
An unanswered station becomes available after it is used for another call and then hangs
up.
The call-back feature expires after a period of 40 minutes.
Facility and non-facility associated signaling
Facility and non-facility associated signaling allows an ISDN-PRI DS1/E1 interface D-channel to
carry signaling information for B-channels (voice or data). D-Channel Backup can also be
administered to increase system reliability.
Feature plus
Feature plus enables those users without DID service to direct dial users on a remote PBX
through the public network. ISDN feature plus eliminates the need for attendant intervention for
those without DID capabilities.
ISDN-Basic Rate Interface
Enables connection of the system to equipment or endpoints that support an Integrated
Services Digital Network (ISDN) by using a standard format called the Basic Rate Interface
(BRI). This feature is a 192-Kbps interface that carries two 64-Kbps B-channels and one
16-Kbps D-channel.
ISDN is a global access standard that uses a layered protocol. It eliminates the need for
multiple, separate access arrangements for voice, data, facsimile, and video services and
136 Avaya AuraTM Communication Manager Overview
ISDN trunks
networks. Using the same pair of wires that carry simple telephone calls, ISDN can deliver
voice, data, and video services in a digital format.
The ISDN-BRI Trunk circuit pack allows Communication Manager to support the T interface and
the S/T interface as defined by ISDN standards (ITU-T recommendation I.411). The circuit pack
provides eight ports to the network and supports two B channels and one D channel.
The ISDN-BRI Trunk provides the following advantages:
l
Provides an inexpensive way to connect to ISDN services provided by the network
provider
l
Meets almost all ETSI Country protocol requirements
l
Supports essential (not supplementary) ISDN services
BRI trunks support public-network access outside the U.S. on point-to-midpoint connections,
with the restriction that Communication Manager must not be configured in a passive bus
arrangement with other BRI endpoints. ISDN-BRI trunks can be used as inter-PBX tie lines
using the QSIG peer protocol. See Figure 4: Communication Manager and ISDN on page 138.
Issue 6 May 2009
137
Trunk connectivity
Figure 4: Communication Manager and ISDN
7
6
1
10
2
8
?
3
9
8
5
4
4
4
Figure notes:
1.
System running Communication
Manager
6.
Private ISDN (can be carried over
ATM-CES)
2.
System running Communication
Manager
7.
Public ISDN (can be carried over
ATM-CES)
3.
System running Communication
Manager
8.
Public and private networks
4.
Basic rate interface telephone
9.
Central office switch
5.
Passive bus
10.
Tandem switch
Multiple subscriber number - limited
The ISDN standard MSN feature lets customers assign multiple extension to a single BRI
endpoint. The MSN feature works with BRI endpoints that allow the channel ID IE to be
encoded as “preferred.”
138 Avaya AuraTM Communication Manager Overview
Multi-Frequency Packet signaling - Russia
NT interface on TN556C
Communication Manager supports the NT (network) side of the T interface using the TN556C
circuit pack. This gives the switch full tie trunk capability using BRI trunks. Communication
Manager supports leased BRI connections through the public network, with a TN2185 on each
end of the leased connection. Communication Manager will not, however, allow customers to
administer both endpoints and trunks on the same TN556C circuit pack.
Presentation restriction
Restricts the display of calling/connected numbers over ISDN trunks. ISDN trunk groups can be
administered to control the display of calling/connected numbers. Each trunk group can be
administered to display “presentation restricted,” “number no available due to networking,” or an
administered text string instead of the calling/connected number.
Wideband switching
Provides the ability to dedicate two or more ISDN B-channels or DSO endpoints for applications
that require large bandwidth. Certain applications, such as video conferencing and high-speed
data transmission, require extra bandwidth and it becomes necessary to put several ISDN-PRI
narrowband channels into one wideband channel to accommodate the needs of these
applications.
This feature supports both European and North American standards.
Multi-Frequency Packet signaling - Russia
See Multi-Frequency Packet signaling on page 97.
National private networking support - Japan
See National private networking support.
Issue 6 May 2009
139
Trunk connectivity
Personal Central Office Line
Provides a dedicated trunk circuit between multi-appearance telephones and a CO or other
switch via the network.
Release Link Trunks
Release Link Trunks (RLT) are used between switch locations to provide centralized attendant
service or automatic call distribution group availability.
Remote access trunks
Tie trunks
Tie trunks carry communications between Communication Manager and other switches in a
private network. Several types of trunks can be used, depending on the type of private network
you establish.
Timed automatic disconnect for outgoing trunk calls
This feature provides the capability to automatically disconnect an outgoing trunk call after an
administrable amount of time. The amount of time that can elapse before the trunk is dropped
can be specified, and can vary between 2 and 999 minutes. If the timer field is blank (the default
value), the feature is disabled and the trunk will not be automatically disconnected.
Timed call disconnection applies to all outgoing trunk calls initiated by a party belonging to a
specified Class of Restriction (COR).
The outgoing trunk disconnect timer only affects outgoing public network trunks (CO, DIOD, FX,
WATS, and ISDN public-network).
140 Avaya AuraTM Communication Manager Overview
Wide Area Telecommunications Service
Note:
The outgoing trunk disconnect timer should be administered to a value large
enough to provide users with adequate response time.
Note:
The outgoing trunk disconnect timer does not apply to outgoing trunk calls that are emergency
or service calls. Specifically, the outgoing trunk disconnect timer does not apply to calls with
ARS call types alrt, emer, nsvc, op, svcl, svfl, svct, or svft.
The outgoing trunk disconnect timer starts after the outgoing trunk call is answered. The
outgoing trunk call is considered answered if:
l
the network provides an answer supervision line signal
l
an ISDN CONNect message is received
l
the Answer Supervision Timeout timer expires
l
the call classifier classifies the call as answered
l
the Outgoing End of Dial Timer expires
Prior to disconnecting the trunk, warning tones are applied to all parties on the call. The first
warning tone occurs when one minute remains on the call. The second warning tone occurs
when 30 seconds remain on the call.
Wide Area Telecommunications Service
Wide Area Telecommunications Service (WATS) trunks allow you to place long-distance
outgoing voice-grade calls to telephones in defined service areas. The calls are priced
according to distance in the service area, length of the call, time of day, and the day of the week.
Issue 6 May 2009
141
Trunk connectivity
142 Avaya AuraTM Communication Manager Overview
Chapter 13: Public Networking and connectivity
Caller ID on analog trunks
Caller ID on analog trunks allows the system to accept calling name information from a Local
Exchange Carrier (LEC) network that supports the Bellcore calling name specification. The
system can send calling name information in the format if Bellcore calling name ID is
administered.
Caller ID on digital trunks
In the United States, the telephone of a user displays calling party information (if the telephone
is a display telephone). Name and calling number are available from the US central offices.
This feature may be used in countries that comply with either US. The display of name and
number will work with all Communication Manager digital telephones (DCP and BRI) equipped
with a 40-character or a 32-character alphanumeric display.
DS1 trunk service
See DS1 trunk service on page 127.
Echo cancellation - with UDS1 circuit pack
See Echo cancellation - with UDS1 circuit pack on page 128.
E1
See E1 on page 128.
Issue 6 May 2009
143
Public Networking and connectivity
T1
See T1 on page 128.
Flexible billing
See Flexible billing on page 48.
Local exchange trunks
Local exchange trunks connect Communication Manager to a central office. The following local
exchange trunks are some of the types available.
800-service trunks
800-service trunks let your business pay the charges for inbound long-distance calls so that
callers can reach you toll-free.
Central Office trunks
See Central Office on page 133.
Digital Service 1 trunks
See DS1 trunk service on page 127.
Direct Inward Dialing trunks
See Direct Inward Dialing on page 134.
144 Avaya AuraTM Communication Manager Overview
QSIG Supplementary Service - Advice of Charge
Direct Inward/Outward Dialing trunks
See Direct Inward/Outward Dialing on page 134.
Foreign Exchange trunks
See Foreign Exchange on page 134.
Wide Area Telecommunications Service
See Wide Area Telecommunications Service on page 141.
QSIG Supplementary Service - Advice of Charge
The QSIG Supplementary Service - Advice of Charge (SS-AOC) provides the capability to
extend the public network charging information, provided by service providers in various
countries, into users in a private network. This is done by transiting the charge information from
a gateway enterprise system to the end user’s enterprise system and desktop display.
This support is for both the AOC-D (charging information during the call), and the AOC-E
(charging information at the end of the call).
Issue 6 May 2009
145
Public Networking and connectivity
146 Avaya AuraTM Communication Manager Overview
Chapter 14: Intelligent networking
Avaya VoIP Monitoring Manager
See Avaya VoIP Monitoring Manager on page 217.
Distributed Communications System protocol
The Distributed Communications System (DCS) protocol allows you to configure two or more
switches as if they were a single, large system. DCS provides attendant and voice-terminal
features between these switch locations. DCS simplifies dialing procedures and allows
transparent use of some of the Communication Manager features. (Feature transparency
means that features are available to all users on DCS regardless of the switch location.) For
more information, also see Centralized Attendant Service on page 43 and Inter-PBX attendant
calls on page 39.
Attendant with DCS
Direct trunk group selection
See Direct trunk group selection on page 39.
Display
See Display on page 43.
DCS automatic circuit assurance
See DCS automatic circuit assurance on page 222.
Issue 6 May 2009
147
Intelligent networking
DCS over ISDN-PRI D-channel
Enhances DCS by allowing access to the public network for DCS connections between DCS
switch nodes. With this feature (also known as DCS Plus or DCS+), DCS features are no longer
restricted to private facilities. The ISDN-PRI B-channel is used for voice communications, and
the ISDN-PRI D-channel is used to transport DCS control information.
DCS protocol - Italy
See Distributed Communications Systems protocol on page 96.
DCS with reroute
Allows a DCS call to be rerouted over a different path if the switch finds a better quality and
lower cost route. This feature allows for rerouting the call after a transfer or rerouting during a
call. DCS with reroute is similar to the rerouting capabilities used with QSIG.
QSIG/DCS voice mail interworking
See QSIG/DCS voice mail interworking on page 110.
Electronic Tandem Network
In an Electronic Tandem Network (ETN) - also known as Private Network Access
(PNA) - Communication Manager provides a variety of features on a network-wide basis. It
allows calls to other systems in a private network. These calls do not use the public network.
Instead, they are routed over your dedicated facilities.
Automatic alternate conditional routing
You can control the routing of particular calls using conditional routing. For example, you can
limit the number of communications satellite hops (communications satellite links used as
trunks) in any end-to-end private network routing pattern. Limiting the number of satellite hops
may be desirable for controlling transmission quality or call delay in both voice and data calls.
148 Avaya AuraTM Communication Manager Overview
Extension number portability
Trunk signaling and error recovery
The reliability of electronic tandem network calls is improved by allowing a trunk call to be
retried on another circuit when signaling failures occur.
l
l
l
tandem switch: A switch within an ETN that provides the logic to determine the best route
for a network call, possibly modifies the digits outpulsed, and allows or denies certain calls
to certain users.
tandem through: The switched connection of an incoming trunk to an outgoing trunk
without human intervention.
Tandem Tie-Trunk Network (TTTN): A private network that interconnects several
customer switching systems.
See also Port network and gateway connectivity on page 121.
Extension number portability
When employees move within the network, they can retain their extension numbers. The ability
to keep extension numbers, and even electronic tandem network and direct inward dialed
numbers, when moving to other locations within the company eliminates missed calls and saves
valuable time.
Internet Protocol
The capabilities and applications of Communication Manager are extended using IP.
Communication Manager IP supports audio/voice over a LAN or WAN, and it ensures that
remote workers have access to communication system features from their PCs. Communication
Manager also provides standards based control between Avaya 8XXX Server and media
gateways allowing communications infrastructure to be distributed to the edge of the network.
The Communication Manager IP engine offers features that enables users to increase the
quality of voice communications. The Quality of Service (QoS) feature enables users to
administer and download the differentiated services type-of-service value to optimize voice
quality. The QoS feature reduces latency by implementing buffers in the audio-processing
board, and assists some routers in prioritizing audio traffic.
Communication Manager IP also includes hairpin and IP-IP direct connections, two features
that make voice communications more efficient. These features increase the efficiency of voice
communications by reducing both per port costs and IP bandwidth usage.
Issue 6 May 2009
149
Intelligent networking
IP solutions supports trunks, IP communications devices, IP port networks, and IP control for
media gateways. IP solutions is implemented using various IP-media processor circuit packs
inside the servers or the Avaya media gateways. The IP media processors provides H.323 trunk
connections and H.323 voice processing for IP telephones. The features that use the IP media
processor also require the CLAN circuit pack or native processor ethernet connectivity.
The IP LAN can also connect through VPN and WAN facilities to extend the customer IP
network across geographically disparate locations. Distributed communication services (DCS+),
or QSIG services, can extend feature transparency, centralized voice mail, centralized
attendant service, call center applications, and enhanced call routing across IP trunks.
Note:
Note:
To maximize voice quality using IP, you must consider both your hardware and
network configurations. For example, with IP softphones, you can send the audio
over traditional circuit switch lines, providing high quality voice, or over IP using
LAN connections. The IP network must be a switched ethernet infrastructure and
have the appropriate engineering to accommodate bandwidth, latency and
packet loss requirements to effectively provide for real-time voice over IP traffic.
Alternate gatekeeper and registration addresses
When an IP endpoint (including softphones, IP telephones, and Avaya R300) registers with the
switch, the switch sends back an IP registration address. The switch sends a different IP
address for each registration according to a cyclic algorithm.
If registration with the original CLAN circuit pack IP address is successful, the switch sends
back the IP addresses of all the CLAN circuit packs in one network region, not including
interconnected regions. These CLAN addresses are called gatekeeper addresses. These
addresses can also be used if the call signaling on the original CLAN circuit pack fails.
Note:
Note:
On switches using the LAN region based on IP Address feature, it is likely that
the network region number assigned to an IP telephone would be different from
the network region number of the TN799 that the telephone is registering
through. That difference would mean the list of TN799 addresses in the same
network region as the IP telephone would be empty. The alternate gatekeeper
feature would send a blank list to the IP telephone.
To prevent that from happening, an IP terminal registers with Communication
Manager. Communication Manager then sends to the endpoint the IP addresses
of the CLANs in the same region as the terminal, followed by network regions
interconnected with the network region of the terminal.
If the network connection to one CLAN circuit pack fails, the IP endpoint re-registers with a
different CLAN. Alternate gatekeeper and registration addresses, and CLAN circuit pack load
sharing, spread IP endpoint registration across more than one CLAN circuit pack, increasing
performance and reliability.
150 Avaya AuraTM Communication Manager Overview
Internet Protocol
Classless Interdomain Routing
Classless Interdomain Routing (CIDR) is a redefinition of the subnet mask, allowing for the
aggregation of contiguous classful networks under a single network definition. This allows for
more efficient routing table management when administering IP address on Communication
Manager.
Multiple network regions per CLAN
Multiple network regions per CLAN enables a single CLAN to provide registration and call
control to IP endpoints in multiple network regions. Communication Manager implements this
approach by allowing IP addresses to be mapped to network regions in a mapping screen,
instead of just to a CLAN. When an IP telephone registers, the switch determines the
telephone's network region number based on the telephone's IP address.
Multiple location support for network regions
Multiple location support for network regions allows remote Avaya media gateways connected
to a central Avaya 8XXX Server to retain:
l
Local user time
l
Local ARS public analysis tables for local trunking
l
Automatic daylight savings time
l
Local touch tone receivers for IP communications devices, such as Avaya IP telephones.
Communication Manager allows administrators to map locations to IP network regions.
Daylight Savings Time rules change
Communication Manager is updated to comply with the new Daylight Saving Time (DST) in the
U.S. DST is extended beginning in 2007. DST now starts the second Sunday of March, and
ends on the first Sunday of November.
Network regions
Network regions provide the administrative foundation on which Communication Manager
features are allocated to IP endpoints. A network region is a collection of IP endpoints and
switch IP interfaces interconnected by an IP network.
Issue 6 May 2009
151
Intelligent networking
Endpoints that share network regions typically represent users with common interests. For
example, a customer might have two separate small campuses in a large metropolitan area,
interconnected by a WAN, and both served by the same server running Communication
Manager. Communication Manager allows the customer to define a network region for each
campus, and associate each of their CLAN and IP media processor circuit packs with these
regions.
Processor Ethernet
The Processor Ethernet interface is one way of connecting VoIP and IP-based devices to
Communication Manager. The Processor Ethernet label is the representation of the computer’s
native ethernet interface inside of the Communication Manager application.
An Enterprise Survivable Server (ESS) and a Local Survivable Processor (LSP) registers with
the main server when it is first configured, and every time it receives a file sync from the main
server. You do not have to administer the Processor Ethernet interface for registration
purposes. The system software enables the use of the Processor Ethernet interface on servers
configured as an LSP or and ESS server.
! CAUTION:
Do not disable the Processor Ethernet interface on an LSP or an ESS server.
Disabling the Processor Ethernet interface disables the LSP or ESS server’s
ability to register with the main server. The LSP or ESS server will not work if the
Processor Ethernet interface is disabled.
CAUTION:
You can administer the server where IP endpoints may register through:
l
The Processor Ethernet interface only
l
CLAN interfaces only (requires the configuration to have CLANs)
l
Either the Processor Ethernet interface or specified CLAN interfaces (requires the
configuration to have CLANs) That is, both interfaces must be able to be enabled at the
same time with some endpoints registering through the Processor Ethernet interface and
some through CLANs.
Adjuncts
The following adjuncts are supported for connectivity to the Processor Ethernet interface:
LSP or Simplex ESS
l
CMS
l
CDR
l
Application Enablement Services (AES)
152 Avaya AuraTM Communication Manager Overview
Internet Protocol
Duplex ESS
l
CDR
l
Messaging
l
SIP Enablement Services (SES)
An adjunct link is established between the LSP or the ESS server. Administration that allows
dedicated and shared connections between the adjuncts and the servers must allow for the link
to remain active at all times. When the LSP or ESS server is not active, the adjunct does not
receive data from that server.
For information on how to a administer the adjunct link, see the adjunct documentation that is
specific to your adjunct.
Merge of IP Connect and Multiconnect configurations
Communication Manager uses the System Management Interface to configure Avaya 8XXX
Servers. You can now choose how to configure each Ethernet Network Interfaces (NIC) based
on their network and policies. This feature applies to all Linux-based Communication Manager
servers that have more than two Ethernet Network Interfaces (NICs).
A customer can assign the following functions to any server NIC:
l
Enterprise LAN
l
Duplication LAN (duplicated servers only)
l
Control Network A
l
Control Network B (duplicated control networks only)
l
Services Laptop
This feature eliminates all reference to, and the distinction between, S87XX Multi-Connect and
IP Connect. Eliminated, too, are all restrictions and rules on Ethernet port assignment except:
l
l
No other functions may be assigned to the NIC used for the Services Laptop.
The duplication LAN must use a gigabit NIC. Avaya highly recommends that no other
functions be assigned to the NIC used for duplication.
H.248 and H.323 registration
The system uses the Processor Ethernet interface to register H.248 gateways and H.323
endpoints. In Communication Manager, Processor Ethernet is supported on simplex and duplex
servers for connection of H.323 devices, H.248 gateways, SIP trunks, and most adjuncts.
Issue 6 May 2009
153
Intelligent networking
S8500 Servers
With the increased functionality of the Processor Ethernet interface, the role of the S8500
Server has been expanded. This expansion includes an S8500 configured as an LSP, and an
S8500 configured as a main server in an IP configuration with no port networks.
Quality of Service
By employing a variety of Quality of Service (QoS) features, Communication Manager provides
the best possible end-to-end audio experience when all or part of the audio path is carried over
packet facilities. “Best” in this context is defined by the customer as represented by the system
administrator, and represents a trade-off between audio reproduction quality, audio path delay
(latency), audio loss, and network resource consumption.
802.1p/Q
IEEE standard 802.1Q and 802.1p provide the means to specify both a Virtual LAN (VLAN) and
a frame priority at layer 2 for use by LAN hubs, or bridges, that can do routing based on MAC
addresses. 802.1p/Q provides for 8 levels of priority (3 bits) and a large number (12 bits) of
VLAN identifiers. The VLAN identifier at layer 2 permits segregation of traffic to reduce traffic on
individual links. Because 802.1p operates at the MAC layer, its presence may vary from LAN
segment to LAN segment within a single network region. Flexibility requires that 802.1p/Q
options be administered individually for each network interface.
Camp-on/Busy-out
A camp-on/busy-out command is commonly used by system technicians to busy-out system
resources that need maintenance or repair. Without it, all active calls using those resources are
indiscriminately dropped if the resource is physically removed from the system. This disruptive
action causes problems for customers, especially when a large number of calls are torn down.
The Camp-on/Busy-out feature for Prowler, MedPRO, and Cruiser adds the ability to remove
idle VoIP resources from the system pool of available VoIP resources.
Note:
Note:
This feature is not supported by the G700 or G350 Media Gateway platforms.
The Camp-on/Busy-out feature enables the user to select the media processor to be busied-out
while the media processor is still in service. After a call ends that was using resources within the
specified media processor, the idled resource is removed from the system pool of available
resources. Once all of the media processor resources are in a “busy-out” state, the associated
board can be removed from the system without disrupting active calls.
154 Avaya AuraTM Communication Manager Overview
Internet Protocol
Call Admission Control bandwidth management
In order to ensure Quality of Service for Voice over IP calls, there is a need to limit overall VoIP
traffic on WAN links. The Call Admission Control (CAC) Bandwidth Management feature of
Communication Manager allows the customer to specify a VoIP bandwidth limit between any
pair of IP network regions. The feature then denies calls that need to be carried over the WAN
link that exceed that bandwidth limit.
CLAN load balancing
CLAN load balancing is the process of registering IP endpoints to CLAN circuit packs (TN799x).
Load balancing occurs among CLANs within a network region.
IP endpoint registration among CLAN circuit packs is done through an algorithm. This algorithm
tracks the number of sockets being used per TN799x circuit pack, and registers IP endpoints to
the TN799x with the most available (unused) sockets. This algorithm applies to H.248, H.323
signaling groups, H.323 stations, and SIP endpoints. Sockets used by adjuncts are not included
in the socket count.
Codecs
A codec (coder/decoder) provides the means by which audio is compressed. A codec is
typically used in VoIP. Some of the codecs supported by Communication Manager include
G.711, G.722, G.723, and G.729.
Differentiated services
With the Differentiated Services (DiffServ) option, the system administrator can administer (by
region) and download, to the TN2302AP, the DiffServ Type-Of-Service (TOS) value. This allows
data networking equipment to prioritize the audio stream at the IP level to promote voice quality.
DiffServ makes use of the TOS octet in the existing IP version 4 header. As such, it may be set
by information senders and used by IP (layer 3) routers within the network.
Dynamic jitter buffers
Propagation delay and jitter is caused when a human's voice is sampled, encoded, and
packetized for transport over the IP network, but is received and decoded at different rates.
Jitter buffers are used to buffer the audio output to smooth the conversions. Communication
Manager provides dynamic jitter buffers to balance both delay of conversation and rapid bursts
that may occur.
Integration with Cajun rules
Cajun rules provide a central repository for QoS parameters and allows comprehensive QoS
management across routers, switches, and endpoints. QoS parameters and policies are
Issue 6 May 2009
155
Intelligent networking
assigned according to network regions on a network region and are distributed through
enterprise directory gateway to Communication Manager and to routers and switching devices.
IP overload control
This enhancement more effectively manages processor occupancy overload situations. The
enhancement applies selected overload mechanisms at a lower occupancy threshold in an
effort to avoid more serious symptoms experienced at higher occupancy levels.
The IP overload control enhancement:
l
l
l
l
l
fortifies the system against bursts of registration traffic
provides a mechanism to alert the far-end to abstain from issuing registrations for some
specified period of time
records the event to maintain a history of potential performance problems
optimizes the maximum number of simultaneous registrations the server can handle based
on the available memory and CPU cycles
reduces the frequency that a server might go into overload due to network problems
IPSI administration enhancements
The TN2312BP IP server Interface (IPSI) administration enhancements enable you to
administer and manage Quality of Service (QoS) and Ethernet interface settings parameters
using the SAT interface. You can administer and manage certain IPSI related parameters such
as the gateway IP address and subnet mask. QoS settings are standardized to communicate
between the IPSI and the Communication Manager. Communication Manager generates a
warning alarm if it finds any discrepancies in Quality of Service (QoS) and Ethernet interface
settings parameters. The IPSI administration inter-operates with Communication Manager by
using the pre-existing QoS and administration interface.
Note:
Note:
The initial IPSI settings must still be administered using the command line
interface.
QoS for call control
Communication Manager allows QoS for the signaling packets coming from gatekeepers such
as the CLAN by employing the same standards based DiffServ and 802.1p/Q schemes as with
audio channels. QoS services further improve the users VoIP audio experience.
QoS for VoIP
Communication Manager implements QoS through the selection of audio codec such as G.711,
G.723 and G.729, and by requesting network prioritization through the layer 3 differentiated
156 Avaya AuraTM Communication Manager Overview
Internet Protocol
services (DiffServ) scheme, as well as the layer 2 IEEE 802.1p/Q prioritization. Diffserv and
802.1p/Q are supported on voice packets coming to/from the gateway, all the way down to the
endpoints such as IP telephones. Dynamic jitter buffers are also used.
QoS to endpoints
Users can set operating parameters to optimize the audio performance, or quality of service
(QoS), on calls made over your IP network. These parameters include the audio codec, network
priority through DiffServ capability, and the IEEE 802.1p/Q MAC-layer prioritization and
segregation.
Default QoS parameters are downloaded to the IP telephone R1.5 and the IP softphone R3
when the values are not provided by the endpoint installer or the user. Certain options can be
set locally by the endpoints or through the gatekeeper. The endpoints receive the parameters
when the endpoints register, and once they are registered, whenever the administered values of
the QoS parameters are modified.
Resource Reservation Protocol
Resource Reservation Protocol (RSVP) is a QoS signaling protocol. RSVP provides a means of
specifying the requirements of IP packet flow, and determining if the intervening network can
provide the resources to protect that flow through a process called “admission control.”
RSVP protection of VoIP audio streams on WANs and other links that are susceptible to
congestion can safeguard the quality of VoIP calls already in progress.
l
IP telephones or gateways request the network routers to reserve bandwidth.
l
The routers act upon the request to allocate bandwidth according to the QoS request.
l
When the bandwidth is reserved, the call is protected against other network traffic in a
loaded or congested network, thereby ensuring good voice quality.
Administrators can now configure RSVP settings in Communication Manager. When the RSVP
enable field in the IP Network Region screen is set to y, the RSVP Reservation
Parameters field appears.
Sending and receiving faxes over IP
Starting with Communication Manager release 2.1, users can send and receive faxes over the
voice over IP (VoIP) and modem over IP (MoIP) networks. The firmware that is resident on the
TN2302AP circuit packs (Hardware Vintage 10 or later), the MM760 Media Module, the G700
Media Gateway, and the G350 Media Gateway, actually performs the processing necessary to
allow proper handling of faxes over an IP network.
Issue 6 May 2009
157
Intelligent networking
Modem over IP
The modem over IP (MoIP) feature allows for transport of data over a 64kbps unrestricted clear
channel. Starting with Communication Manager release 3.0, when a clear channel data call is
originated, the system communicates to the media processor or VoIP engine to allow a 64kpbs
clear channel to be opened for transport.
Relay mode
In relay mode, the firmware detects fax tones and uses the appropriate modulation protocols
(V.xx) to terminate or originate the fax so that the fax can be carried over the IP network. To
reduce bandwidth over the IP network, the system encodes the modulated analog signal from
the fax, and uses a relay coder/decoder. This process improves the reliability of transmission.
Also, because the data packets for faxes in relay mode are sent almost exclusively in one
direction, from the sending endpoint to the receiving endpoint, bandwidth use is reduced.
Relay mode works only if the receiving fax endpoint and the sending fax endpoint both
communicate through Avaya 8XXX Servers. This transport of fax signals occurs at a 9600 bps
rate (though this rate may vary with the version of firmware). This mode may be used for fax
calls to and from Communication Manager R2.0 systems.
Pass through mode
Alternatively, you can choose to have fax signals sent in “pass through” mode. Pass through
mode means the fax signals are transported using G.711-like encoding and are delivered to the
receiving fax endpoint as IP signals. This capability provides higher quality transmission in the
circumstance where endpoints in the network are all synchronized to the same clock source.
Pass through mode works only if the receiving fax endpoint and the sending fax endpoint both
communicate through Avaya 8XXX Servers.
The transport speed is up to the equivalent of circuit-switched calls and supports G3 and Super
G3 fax rates, up to and including 33.6 kbps.
! CAUTION:
CAUTION:
If users are using Super G3 fax machines as well as modems, do not assign
these fax machines to a network region with an IP Codec set that is
modem-enabled as well as fax-enabled. If its Codec set is enabled for both
modem and fax signaling, a Super G3 fax machine incorrectly tries to use the
modem transmission instead of the fax transmission.
Therefore, assign modem endpoints to a network region that uses a
modem-enabled IP Codec set, and assign the Super G3 fax machines to a
network region that uses a fax-enabled IP Codec set.
You can assign packet redundancy in both pass through and relay mode, which means the
media gateways use RFC 2198 packet redundancy to improve packet delivery and robustness
of fax transport over the network.
158 Avaya AuraTM Communication Manager Overview
Internet Protocol
Pass through mode uses more network bandwidth than relay mode. Redundancy increases
bandwidth usage even more.
Encryption
You can encrypt fax pass through calls using either Avaya Encryption Algorithm (AEA) or
Advanced Encryption Standard (AES). You can encrypt fax relay calls with AEA only. For more
information about encryption, see Security, privacy, and safety on page 197.
T.38 faxes over the Internet
The users can send and receive faxes over the VoIP network using the T.38 standard for relay.
The firmware resident on the TN2302AP circuit packs (Hardware Vintage 10 or later), the
MM760 Media Module, the G700 Media Gateway, and the G350 Media Gateway actually
performs the processing necessary to allow proper handling of faxes over an IP network. This
transport of fax signals occurs at a 9600 bps rate.
The T.38 fax capability allows users to send faxes to and receive faxes from endpoints that are
connected to non-Avaya systems. This capability is standards-based and uses IP trunks and
H.323 signaling to allow communication with non-Avaya systems. Additionally, the T.38 fax
capability uses the UDP protocol.
Note:
Note:
Fax endpoints served by two different Avaya 8XXX Servers can also send T.38
faxes to each other if both systems are enabled for T.38 fax. In this case, the
servers also use IP trunks.
However, if the T.38 fax sending and receiving endpoints are on port networks or
media gateways that are registered to the same server, the gateways or port
networks revert to Avaya fax relay mode. Avaya fax relay mode is more efficient
that T.38 from a bandwidth perspective.
Both the sending and receiving systems must announce support of T.38 fax data applications
during the H.245 capabilities exchange. Avaya systems announce support of T.38 fax if the
capability is administered on the Codec Set screen for the region and a T.38-capable media
processor was chosen for the voice channel. In addition, for a successful fax transmission, both
systems should support the H.245 null capability exchange (shuffling) in order to avoid multiple
IP hops in the connection.
Note:
Note:
The T.38 fax capability does not support TCP.
You can assign packet redundancy to T.38 standard faxes to improve packet delivery and
robustness of fax transport over the network.
Issue 6 May 2009
159
Intelligent networking
Pass through mode
You cannot send faxes in pass through mode with the T.38 standard.
Shuffling and hairpinning
Shuffling and hairpinning can improve traffic handling performance and improve voice quality by
more efficiently using both Communication Manager switching fabric by allocating, when
possible, available IP network resources.
“Shuffling” means rerouting the audio channel connecting two IP endpoints. After shuffling, the
audio which previously was carried in a mixed connection of IP signaling and TDM bus
signaling, goes directly through the LAN or WAN between the two IP endpoints. Shuffling also
can mean reversing this process if an endpoint requests a resource to support a feature, such
as conferencing that requires the TDM bus.
“Hairpinning” means rerouting the audio channel connecting two IP endpoints so that the bearer
(audio) packets are routed through the TN2302AP IP Media Processor board in IP format,
without having to go through the IP to TDM conversion or traverse the TDM bus.
G.722 shuffling over H.323/SIP trunks
During a typical phone call that goes over the PSTN, it is sometimes difficult to distinguish
certain letter sounds, such as “f” and “s” or “t” and “d.” The reason for this is that most of the
speech energy that allows humans to distinguish consonants occurs above 3,000 Hz. The best
digital PSTN phone call doesn’t pass any energy above 3,300 Hz.
High-fidelity audio (sometimes called wideband audio) passes audio frequencies of up to 7,000
Hz. Research has shown that conversations over these higher frequencies dramatically
improve intelligibility and reduce listener fatigue. They also make it much easier to understand
accented speech.
G.722 is an audio coding system (50 to 7,000 Hz) that can be used in a variety of higher quality
speech applications. This feature supports endpoints capable of handling G.722 codecs, but it is
specifically intended for the Avaya 96xx H.323 telephones and G.722-capable conference
bridge endpoints.
NAT with shuffling
Communication Manager allows IP endpoints to shuffle if they are behind a Network Address
Translation (NAT) device in an IP network.
Note:
Note:
Network Address Translation (NAT) is a method to address the shortage of IP V4
addresses by allowing globally register IP addresses to be reused by native
networks. A NAT device translates between translated and native IP addresses.
160 Avaya AuraTM Communication Manager Overview
Internet Protocol
Communication Manager supports IP direct calls (a call that has been shuffled) between two IP
endpoints that are translated through a NAT device.
This enhancement works with static one-to-one NAT. It does not facilitate Port Address
Translation (PAT), also known as Network Address Port Translation (NAPT). This enhancement
does not work with many-to-one NAT.
TTY
People with hearing or speech disabilities often rely on a device known as a TTY in order to
communicate on telephone systems. The term “TTY” is an abbreviation for Teletypewriter. The
term “TDD” (Telecommunication Device for the Deaf) is also frequently used. The term TTY is
generally preferred, however, because many people who use these devices are not deaf.
TTY devices typically resemble small laptop computers, except that there is a one- or two-line
alphanumeric display in place of the computer screen.
Connection to the telephone network is generally through an acoustic coupler into which the
user places the telephone's handset, or through an analog RJ-11 tip/ring connections.
Reliable transmission of TTY signals is supported by Communication Manager. This complies
with the requirements and guidelines outlined in United States accessibility-related laws. Those
laws include:
l
Titles II, III, and IV of the Americans with Disabilities Act (ADA) of 1990.
l
Sections 251 and 255 of the Telecommunications Act of 1996.
l
Section 508 of the Workforce Investment Act of 1998.
Communication Manager TTY support is currently restricted to TTY devices that use the:
l
l
US English standard TTY protocol, specified by ANSI/TIA/EIA 825 as: “A 45.45 Baud FSK
modem.”
UK English standard TTY protocol, Baudot 50.
Important characteristics of this standard are:
l
Note:
TTYs are silent when not transmitting. Unlike fax machines and computer modems, TTYs
have no “handshake” procedure at the start of a call, nor do they have a carrier tone during
the call. This approach has the advantage of permitting TTY tones, DTMF, and voice to be
intermixed on the same call.
Note:
A large percentage of people who use TTY devices intermix voice and typed TTY
data on the same call. The most common usage is by people who are hard of
hearing, but nevertheless able to speak clearly. These people often prefer to
receive text on their TTY device and then speak in response. This process is
referred to as Voice Carry Over (VCO).
Issue 6 May 2009
161
Intelligent networking
l
l
Operation is “half duplex.” TTY users must take turns transmitting and typically cannot
interrupt each other. If two people try to type at the same time, their TTY devices might
show no text at all or show text that is unrecognizable. Also, there is no automatic
mechanism that lets TTY users know when a character they have correctly typed has been
received incorrectly.
Each TTY character consists of a sequence of seven individual tones. The first tone is
always a “start tone” at 1800 Hz. This is followed by a series of five tones, at either 1400 or
1800 Hz, which specify the character. The final tone in the sequence is always a “stop
tone” at 1400 Hz. The stop tone is a border that separates this character from the next.
The following types of systems support TTY communication:
l
Analog telephones and trunks
l
Digital telephones and trunks
l
VoIP gateways
l
Messaging systems
l
Automated attendant systems
l
IVR systems
l
Wireless systems in which a TTY-compatible coder is used
As long as the user's TTY device supports the following, Communication Manager allows:
l
l
l
Voice and TTY tones to be intermixed on the same call.
DTMF and TTY (with or without voice) to be intermixed on the same call. This allows TTY
users to access DTMF-based voice mail, auto-attendant, and IVR systems.
The use of acoustically coupled and “direct connect” (RJ-11) TTY devices.
TTY over analog and digital trunks
Communication Manager supports TTY calls within a gateway or port network between two
analog telephones. TTY calls from a gateway or port network over analog trunks or digital
trunks is also supported.
TTY over Avaya IP trunks
Communication Manager supports calls over IP trunks, as well as Inter-Gateway Calls (IGC).
Note:
Note:
For this feature to work, both the sender (near end) and the receiver (far end) of a
TTY call must each be connected to Avaya IP trunks. This feature does not work
if either telephone is an IP telephone.
162 Avaya AuraTM Communication Manager Overview
Internet Protocol
TTY relay mode
In relay mode, the system:
l
detects TTY characters
l
transports a representation of the characters over the IP network
l
regenerates TTY characters/tones for delivery to the TTY device
This transport of TTY supports US English TTY (Baudot 45.45) and UK English TTY (Baudot
50). TTY uses RFC 2833 or RFC 2198 style packets to transport TTY characters.
Depending on the presence of TTY characters on a call, the transmission toggles between
voice mode and TTY mode. The system uses up to 16 kbps of bandwidth when sending TTY
characters, and normal bandwidth of the audio codec for voice mode. This mode may be used
for TTY calls to and from Communication Manager R2.0 systems.
In relay mode, you can also assign packet redundancy. Packet redundancy means the media
gateways send duplicated TTY packets to ensure and improve quality over the network.
TTY pass through mode
Alternatively, you can choose to have TTY signals sent in pass through mode. With pass
through mode enabled, when the system detects TTY characters, the system uses G.711
encoding to transport the TTY signals end-to-end over the IP network. G.711 encoding pass
through mode means the TTY signals are changed to digital format, and are delivered to the
receiving endpoint after unencoding the digital signals.
Pass through mode provides higher quality transmission when endpoints in the network are all
synchronized to the same clock source.
In pass through mode, you can also assign packet redundancy. Packet redundancy means the
media gateways send duplicated TTY packets to ensure and improve quality over the network.
Pass through mode uses more network bandwidth than relay mode. Pass through TTY uses
87-110 kbps, depending on the packet size, whereas TTY relay uses, at most, the bandwidth of
the configured audio codec. Redundancy increases bandwidth usage even more.
Variable length ping
Provides an enhancement to the ping command included in R7.1. This enhancement specifies
a longer packet to be sent by ping and shows if a router or host has a problem fragmenting or
integrating transferred packets.
Issue 6 May 2009
163
Intelligent networking
Variable Length Subnet Mask
Variable Length Subnet Mask (VSLM) is a redefinition of the subnet mask, allowing for a more
efficient allocation of IP addresses within a traditional classful block when administering IP
address on Communication Manager.
QSIG
Auto callback - QSIG Call Completion
Auto Callback covers auto callback within a private corporate network only through QSIG. Auto
Callback provides an administrable option on the Trunk Group screen to allow users to specify
the method of signaling connection the system uses while waiting for a busy station to become
idle.
Basic
QSIG provides compliance to the International Standardization Organization (ISO) ISDN-PRI
private-networking specifications. QSIG is defined by ISO as the worldwide standard for private
networks. QSIG features are supported on BRI trunks.
QSIG is the generic name for a family of signaling protocols. The Q-reference point or interface
is the logical point where signaling is passed between 2 peer entities in a private network. QSIG
signaling can provide feature transparency in a single-vendor or multi-vendor environment.
QSIG provides call-related supplementary services. These are services that go beyond voice
or data connectivity and number transport and display. Examples of supplementary services
include name identification, call forwarding (diversion), and call transfer.
Call completion
Call completion utilizes the QSIG platform enhancement call independent signaling connections
and is functionally equivalent to the Distributed Communications System (DCS) feature:
auto-callback. The call completion feature includes a connection release method. The
connection release method clears the Temporary Signaling Connection (TSC) after each phase
of call-independent signaling and establishes a new TSC for each subsequent phase.
164 Avaya AuraTM Communication Manager Overview
QSIG
Call forwarding (diversion)
QSIG call forwarding (diversion) is based on the Communication Manager call forwarding
feature. It extends the feature transparency aspects of call forwarding over a QSIG trunk:
l
l
l
If QSIG call forwarding is activated, all calls are diverted immediately.
If QSIG call forwarding with busy/do not answer is activated and a station is busy, a call is
diverted immediately.
If QSIG call forwarding with busy/do not answer is activated and a station is idle but the
call is not answered, a call is diverted after a specified number of rings.
These features are activated either by dialing a Feature Access Code (FAC) or by pressing a
button.
Issue 6 May 2009
165
Intelligent networking
Call Independent Signaling Connections
Call Independent Signaling Connections (CISC) are used to pass QSIG supplementary service
information that is independent of an active call between two QSIG compliant nodes.
Implementation is based on the ISO standard for CISC. It is possible to determine the status of
a QSIG TSC by using the “status signaling group” command on the SAT.
Call offer
This feature, on request from the calling-user (or on behalf of that user), enables a call to:
l
l
Be offered to a busy called-user
Wait for a busy called-user to accept the call when the necessary resources have become
available
Call transfer
QSIG call transfer differs from the standard Communication Manager transfer feature in that
additional call information is available for the connected parties after the transfer completes.
However, the information is only sent for QSIG trunks. If one call is local to the transferring
switch, that user receives the name of the party at the far end.
Name display on unsupervised transfer
Station A calls station B, and is then transferred to station C. Station B has established a
connection through QSIG to station A, and an enquire connection (intern or through QSIG) to
station C.
Station B transfers station C to station A. The name of station C (not answered enquire call) is
sent after call transfer to station A when the Send Name field on the Trunk Group screen is set
to y at the QSIG trunk side to the primary PINX. So the name of the transferred-to station C
appears on the display of the station A in the primary connection immediately after the call
transfer.
Called name ID
The QSIG called name feature presents the name of the called party on the display of the
calling party while the call is ringing. It then reverts to “connected name” when answered.
166 Avaya AuraTM Communication Manager Overview
QSIG
Centralized Attendant Service
Provides you with the capability to have all your attendants in one location, serving users in
multiple locations. QSIG CAS does not utilize separate Release Link Trunks (RLT). This feature
will not restrict calls from going out over non-QSIG trunks; however, the full functionality of the
QSIG CAS will not be available.
Attendant display of Class of Restriction
While on a call, the attendant can press a “COR display” button to see the class of restriction of
the user. The attendant will not block the transfer of the restricted line to the user. This feature is
used for informational purposes only.
Attendant return call
If a call that is transferred by the attendant goes unanswered for a specified period of time, the
call is returned to the attendant. Preferably the call will transfer back to the attendant who
transferred the call.
Priority queue
QSIG MSI will pass more information to the main PBX. This information enables calls coming in
from a QSIG CAS branch to be placed in the appropriate place in the queue, as if the call
originated on the main PBX.
RLT emulation through a PRI
ISDN QSIG trunks will route calls from the branch PBX to the main PBX. You no longer have to
specify a dedicated RLT network. The QSIG path replacement takes care of the trunk
optimization. You have the flexibility to route calls to the main PBX.
Communication Manager/Octel QSIG integration
Communication Manager enables integration of Octel messaging servers through QSIG. See
Octel integration on page 109.
Complex private numbering plan support
Additional flexibility is provided for private-network numbering in support of customers’ private
networks.
Issue 6 May 2009
167
Intelligent networking
Leave Word Calling
See Leave Word Calling on page 107.
Manufacturer-Specific Information
QSIG handles non-standardized information that is specific to a particular PBX or network. This
information is known as Manufacturer Specific Information (MSI). A manufacturer can define
manufacturer-specific supplementary services operations after it has:
l
l
Applied to a sponsoring and issuing organization (ECMA or European Computer
Manufacturers Association in this case)
Been assigned an organization identifier. This organization identifier is used as the root of
the manufacturer-specific service-operation value.
All MSI operation values should be unique to that manufacturer.
Manufacturer-specific supplementary services can be created using specific operations
encoded with the identifier of the manufacturer. Communication Manager supports non-QSIG
applications that transport information across QSIG networks in MSI. Applications have the
same functionality over QSIG networks that they have over non-QSIG networks. Applications
that use MSI include Centralized Attendant Service, Transfer to AUDIX, Best Service Routing,
and QSIG VALU.
Message Waiting Indication
The system indicates that a guest telephone has received one or more messages in their voice
mailbox. An automatic message waiting lamp light at the telephone of the called party.
Name and number identification
Allows a switch to send and receive the calling number, calling name, connected number, and
connected name. Additional parameters that control the display of the connected name and
number are administered on the Feature-Related System-Parameters screen. QSIG Name
and Number Identification displays up to 15 characters for the calling and connected name and
up to 15 digits for the calling and connected number across ISDN-PRI interfaces.
168 Avaya AuraTM Communication Manager Overview
QSIG
Path replacement with path retention
With this feature, a call between switches in a private network can be replaced with new
connections while the call is active. This feature is invoked when a call is transferred and
improvements may be made in costs.
For example, after a call is transferred, the two parties on the transferred call can be connected
directly and the unnecessary trunks are dropped off the call. The routing administered at the
endpoints may allow for a more cost-effective connection.
Earlier versions of DEFINITY could not route a call over the original trunk when path
replacement was turned on. Path Replacement features Path Retention, which allows
Communication Manager to use the original trunk group path when the routing analysis
performed by the switch shows the original trunk group to be the best route.
QSIG/DCS voice mail interworking
QSIG/DCS Voice Mail Interworking is an enhancement to the current QSIG feature. It integrates
DCS and QSIG Centralized Voicemail through the DCS+/QSIG gateway. Switches labeled
DCS+/QSIG integrate multi-vendor PBXs into a single voice messaging system. QSIG/DCS
Voice Mail Interworking works on G3r, G3si, and G3csi. It provides network flexibility, DCS
functionality without a dedicated T1.
Reroute after diversion to voice mail
Supports path optimization for calls that are diverted to a QSIG voice mail hunt group. That is,
the switch moves the call to the shortest route between the caller and the voice mail system. For
example, if user A on switch A calls user B on switch B and the call goes to a voice mail system
attached to switch C, then the call is using up two trunks: A-B and B-C. If there is a trunk that
directly connects switches A and C, this feature will drop the A-B and B-C connection and set up
a new call from switch A to switch C, thus saving one trunk. The reroute happens automatically;
the user never knows that the extra trunk was dropped.
Stand-alone path replacement
Path Replacement is the process of routing an established call over a more efficient path, after
which the old call is torn down leaving those resources free. Path Replacement offers potential
savings by routing calls more efficiently, saving resources and trunk usage.
Issue 6 May 2009
169
Intelligent networking
Path replacement can exist as a stand-alone feature, or occur in the following additional cases:
l
l
Call Forwarding by Forward Switching supplementary service, including the case where
Call Diversion by Rerouting fails, and Call Forwarding is accomplished via forward
switching
Gateway scenarios where Communication Manager, serving as an incoming or outgoing
gateway, invokes PR to optimize the path between the gateways
l
Calls in queue/vector processing even though no true user is on the call yet
l
QSIG Lookahead Interflow call, Best Service Route call, or adjunct route
Supplementary services and rerouting
The QSIG standard defines Supplementary Services as those service beyond voice or data
connectivity and number transport and display. Examples include call forwarding, transfer and
call hold.
VALU
Call coverage
This feature provides similar call coverage as DCS call coverage and Call Coverage Remote
Off Net (C-CRON). The call will come back if covered over QSIG. The functionality will only be
complete when all the switches are running under Communication Manager and using QSIG
VALU. The covered-to party can still receive distinct alerting.
Call coverage and CAS
When a trunk has both CAS and VALU Call Coverage activated, the coverage display
information is provided on calls that cover from a branch PBX to the main PBX. Path
replacement will be attempted after coverage.
Distinctive alerting
Provides distinctive ringing, internal and external, to the remote called party when the call is
routed over the QSIG network.
170 Avaya AuraTM Communication Manager Overview
Uniform Dial Plan
Uniform Dial Plan
A unique four- or five-digit number assigned to each station on the network. Uniform numbering
gives each station a unique number (location code plus extension) that can be used at any
location in the electronic tandem network to access that station, Communication Manager
enhances the standard UDP with the unrestricted 5-digit Uniform Dial Plan, which allows up to
five digits to be parsed for call routing.
Dial Plan Expansion
With Communication Manager release 4.0, you can expand your dial plan to a maximum of
13 digits. This affects telephones, data modules, login IDs, and vectors.
Administrators have the flexibility to administer dial plans between 3 and 13 digits in length.
Communication Manager supports mixed digit lengths in the same dial plan within a location
and across a network of locations.
Customers upgrading to Communication Manager 4.0 can choose to migrate to the expanded
13-digit dial plan. Customers who choose not to migrate may convert their dial plans at a later
date.
Distributed Communications System (DCS) protocol is limited to a dial plan of 3-5 digits. If your
dial plan requires 6 or 7 digits, QSIG, which is the generic name for a family of signaling
protocols, is required.
Note:
Note:
Some features cannot expand to 13 digits. For a complete description of the Dial
Plan Expansion feature, see the following documents:
l
l
Avaya Aura™ Communication Manager Feature Description and Implementation,
555-245-205.
Administering Avaya Aura™ Communication Manager, 03-300509.
Multi-location dial plans
When a customer migrates from a multiple voice server QSIG/DCS network to a single voice
server whose gateways are distributed across a data network, it may initially seem as if some
dial plan functions are no longer available.
Issue 6 May 2009
171
Intelligent networking
This feature preserves dial plan uniqueness for extensions and attendants that were provided in
a multiple QSIG/DCS network, but were lost when customers migrated to a single distributed
network. This feature provides dial plan capabilities similar to those provided before the
migration, including:
l
extension uniqueness
l
announcement per location
l
local attendant access
l
local ARS code administration
A major reason to migrate customers from a multiple QSIG/DCS environment to a single S87XX
network is to provide a greater set of features and help reduce costs. Migrating to a single
network reduces the number of systems a customer has to maintain. That in turn lowers
administration costs - one switch to administer instead of multiple switches, one dial plan to
maintain, and so on. With a single distributed network solution, some features no longer work
transparently across multiple locations.
For example, in a department store with many locations, each location might have had its own
switch with a QSIG/DCS network. That way, the same extension could be used to represent a
unique department in all stores. For example, extension 123 might be the luggage department
in all stores. If the customer migrates to a single distributed network, this functionality is not
available without this feature.
In addition, an S87XX solution does not assure that a call that is routed to an attendant would
terminate at the local attendant. Let us use an example of a public school district that previously
was networked with a switch at each school. If the school district migrates to an S87XX
network, dialing the attendant access code at your school may not route your call to the local
attendant.
Instead of having to dial a complete extension, the multi-location dial plan feature allows a user
to dial a shorted version of the extension. For example, a customer can continue to dial 4567
instead of having to dial 123-4567. Communication Manager takes the location prefix and adds
those digits to the front of the dialed number. The switch then analyzes the entire dialed string
and routes the call based on the administration on the Dial Plan Parameters screen.
Note:
Note:
You can use the Per-Location Dial Plan feature to allow different branches to
have different short extensions, so that the extensions do not conflict across
branches.
Punctuation on station displays
On digital telephone displays, Communication Manager can display punctuation to make
reading a longer extension easier.
172 Avaya AuraTM Communication Manager Overview
Uniform Dial Plan
Punctuation marks that are allowed include:
l
hyphen (for example, xxx-xxxx)
l
period (for example, xxx.xxxx)
l
space (for example, xx xx xx)
Formats for displaying numbers with punctuation are on the Dial Plan Parameters screen.
For more information on the Dial Plan Parameters screen, see Administering Avaya Aura™
Communication Manager, 03-300509.
Extended trunk access
Used with Uniform Dial Plan, allows the system to send any unrecognized number (such as an
extension not administered locally) to another system for analysis and routing. Such
unrecognized numbers can be Facility Access Codes, Trunk Access Codes, or extensions that
are not in the Uniform Dial Plan table. Non-Uniform Dial Plan numbers are administered on
either the First Digit Table (on the Dial Plan Record screen) or the Second Digit Table. They are
not administered on the Extended Trunk Access Call Screening Table. Extended Trunk Access
helps you make full use of automatic routing and Uniform Dial Plan.
Extension Number Portability - When employees move within the network, they can retain their
extension numbers. The ability to keep extension numbers, and even Electronic Tandem
Network and Direct Inward Dialed numbers, when moving to other locations within the company
eliminates missed calls and saves valuable time.
Issue 6 May 2009
173
Intelligent networking
174 Avaya AuraTM Communication Manager Overview
Chapter 15: Data interfaces
Administered connections
Automatically establishes an end-to-end connection between two access or data endpoints
based on administered attributes. This feature provides capabilities such as alarm notification,
including an administrable alarm type and threshold; automatic restoration of connections
established over a Software-Defined Data Network; ISDN-PRI trunk group [service may be
referred to as ISDN-PRI (AC/AE) Service]; scheduled as well as continuous connections; and
administrable-retry interval for failed connection attempts.
Data call setup
Enables the setting up of data calls using a variety of methods, such as: keyboard dialing,
telephone dialing, Hayes command dialing, permanent switched connections, administered
connections, automatic calling unit interface, and Hot Line dialing. Data Call Setup is provided
for both DCP and ISDN-BRI telephones.
Data hot line
Provides for automatic placement of a data call when the originator hangs up. Data Hot Line
may be used for security purposes. This feature offers fast and accurate call placement to
commonly called data endpoints. Data terminal users who constantly call the same number can
use Data Hot Line to automatically place the call when they hang up the telephone.
Data privacy
Data Privacy protects analog data calls from being disturbed by any overriding or ringing
features of the system. Data Privacy is activated when you dial an activation code at the
beginning of the call.
Issue 6 May 2009
175
Data interfaces
Data restriction
Protects analog data calls from being disturbed by any overriding or ringing features of the
system. It is administered at the system level to selected analog and multi-appearance
telephones and trunk groups.
Default dialing
Provides data terminal users who dial a specific number the majority of the time a very simple
method of dialing that number. This feature enhances Data Terminal (Keyboard) Dialing by
allowing a data terminal user to place a data call to a pre-administered destination in several
different ways, depending on the type of data module. Data Terminal Dialing and Alphanumeric
Dialing are unaffected.
IP asynchronous links
IP asynchronous links enable Communication Manager to transfer existing asynchronous
adjunct connectivity to an Ethernet (TCP/IP) environment. IP asynchronous links support switch
server applications, as well as client applications. Systems running Communication Manager
can connect to System Management applications such as the Avaya Visibility Suite over the
LAN. Call Detail Recording (CDR) devices, Property Management System (PMS) and printers
can be connected using asynchronous TCP/IP links.
IP asynchronous links:
l
l
l
l
l
l
Reduce the cost of connecting to systems running Communication Manager for various
adjuncts
Allow for an open architecture to transport information and increases the speed at which
data is transferred
Allow customers to manage applications from on-site or remote locations
Allow several system management applications to run on a single PC, thereby reducing
hardware requirements
Guarantee data delivery through a reliable session-layer protocol
Support the existing serial hardware investment of a customer through use of Network
Terminal Servers
176 Avaya AuraTM Communication Manager Overview
Multimedia application server interface
Multimedia application server interface
The Multimedia Application Server Interface provides a link between Communication Manager
and one or more Multimedia Communications eXchange nodes. A Multimedia Communications
eXchange is a stand-alone multimedia call processor produced by Avaya. This link to
Communication Manager enhances the capabilities of each Multimedia Communications
eXchange system by enabling it to share some of the Communication Manager features. In
particular, the interface provides the following advantages:
l
l
l
Call Detail Recording (CDR). The capture of call detail records so you can analyze the call
patterns and usage of multimedia calls just as Communication Manager administrators
analyze normal calls.
Automatic Alternate Routing/Automatic Route Selection (AAR/ARS). The intelligent
selection of the most cost-effective routing for calls, based on available resources and your
carrier preference. The system may select public trunks via DEFINITY Multimedia
eXchange (MMCX).
Voice Mail Integration. You can access your EMBEDDED AUDIX or CM Messaging voice
messaging system from a Multimedia Communication eXchange (MMCX).
Multimedia calling
Multimedia calls are initiated with voice and video only. Once a call is established, one of the
parties may initiate an associated data conference to include all of the parties on the call who
are capable of supporting data. The data conference is controlled by an adjunct device called
an Expansion Services Module (ESM).
Multimedia call early answer on vectors and stations
Early Answer is a feature applied to multimedia calls in conjunction with conversion to voice.
Early Answer:
l
Answers the data call
l
Establishes the multimedia protocol prior to completion of a converted call
l
Ensures that a voice path to/from the originator is available when the (voice) call is
answered
For an incoming call, Early Answer answers the dynamic service-link calls when the destination
endpoint answers, unless Early Answer is specified during routing or termination processing.
Issue 6 May 2009
177
Data interfaces
Note:
The “destination voice endpoint” might be an outgoing voice trunk if the
destination voice station is forwarded or covered off-premises.
Note:
Multimedia Call Handling
Multimedia Call Handling (MMCH) enables you to control voice, video, and data transmissions
using your telephone set. The feature buttons on a multi-function telephone enable you to
conduct video conferences, and forward, cover, hold, or park multimedia calls much as you
would a standard voice call. You can also share PC applications so that you and colleagues can
collaborate while working from remote sites. See Figure 5: Multimedia Call Handling
(MMCH) on page 178.
Figure 5: Multimedia Call Handling (MMCH)
9
1
7
3
8
3
2
4
4
5
5
6
cydfmch2 KLC 030102
Figure notes:
1.
One number access
5.
Call redirection
2.
Multimedia call complex
6.
Multimedia conferencing
3.
Multimedia to voice conversion
7.
BRI data connection
4.
Standard voice call handling
8.
DCP voice connection
9.
ESM data collaboration
178 Avaya AuraTM Communication Manager Overview
Pass advice of charge information to world class BRI endpoints
Multimedia call redirection to multimedia endpoint
A dual port multimedia station may be a destination of call redirection features such as call
coverage, forwarding, and station hunting. The station can receive and accept full multimedia
calls or data calls converted to multimedia.
Multimedia data conferencing (T.120) through ESM
The data conference is controlled by an adjunct device called an Expansion Services Module
(ESM). The Expansion Services Module is used to terminate T.120 protocols [including
Generalized Conference Call (GCC), a protocol standard for data conference control] and
provide data conference control and data distribution. The MultiMedia Interface circuit pack,
TN787, is used to rate adapt T.120 data to/from the ESM.
Multimedia hold, conference, transfer, and drop
Station users have the ability to activate hold, conference, transfer, or drop on multimedia calls.
Multimedia endpoints and voice-only stations may participate in the same conference.
Multimedia multiple-port networks
Communication Manager supports the equivalent of 580 Basic mode complexes operating at
6CCS traffic level. All enhanced mode complexes operate with soft-mode service links since the
use of hard-mode service links reduces capacities. G3si limits are 1/3 to 1/2 of the G3r limits,
depending on memory limitations and port network limitations.
Pass advice of charge information to world class BRI
endpoints
Provides Advice of Charge (AOC) information to World Class BRI (WCBRI) endpoints. On a call
using a WCBRI endpoint, AOC information will be displayed on the endpoint after the call has
completed and the far end has hung up.
Issue 6 May 2009
179
Data interfaces
180 Avaya AuraTM Communication Manager Overview
Chapter 16: Call routing
Alternate facility restriction levels
Allows Communication Manager to adjust facility restriction levels or authorization codes for
lines or trunks. Each line or trunk is normally assigned a facility restriction level. With this
feature, Alternate Facility Restriction Levels are also assigned. Attendants can change to the
alternates, thus changing access to lines and trunks. You might want to use this feature to
disable most long-distance calling at night, for example, to prevent unauthorized staff from
making long-distance calls.
! CAUTION:
CAUTION:
This feature may change the AAR and ARS routing preferences. Using it on
tandem and tie-trunk applications affects entire networks. Calls that are part of a
cross-country private network may be blocked.
Automatic routing features
Communication Manager provides a variety of automatic routing features for public and private
networks. Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS) are the
foundation for these automatic-routing features. They route calls based on the preferred
(normally the least expensive) route available at the time the call is placed. Generally, AAR
routes calls over a private network and ARS routes calls using the public network numbering
plan. However, both AAR and ARS support public and private networks. You can use the other
features listed in this section when you use AAR and ARS.
Automatic Alternate Routing
Automatic Alternate Routing (AAR) allows private network calls to originate and terminate at
one or many locations without accessing the public network. When you dial an access code and
telephone number, AAR selects the most desirable route for the call and performs digit
conversion as necessary. If the first choice route is unavailable, another route is chosen
automatically.
The numbers you call using AAR are normally private-network numbers. However, you can call
a public-network number, a service code, an international number, operator access code, or an
operator-assisted dialing number. With AAR and Subnet Trunking, you have a convenient way
Issue 6 May 2009
181
Call routing
to place international calls to frequently-called foreign cities. Such calls route as far as possible
over the private network, and then access the public network. This saves toll charges and
allows you to use your private network as much as possible. In a multi-location system, you can
administer AAR on a per-location basis.
Automatic Route Selection
Automatic Route Selection (ARS) selects carriers automatically and routes calls
inexpensively over the public network. When there are one or more long-distance carriers and
Wide Area Telecommunications Service (WATS) provided, Communication Manager selects the
most preferred route for the call. Long-distance carrier-code dialing is not required on routes
selected by the system. You assign long-distance carrier-codes and Communication Manager
translates them. The system inserts codes as needed to guarantee automatic-carrier selection.
ARS can route calls to a variety of types-of-numbers and access a variety of types of trunk
groups. In a multi-location system, you can administer ARS on a per-location basis.
ARS/AAR dialing without FAC
The Automatic Route Selection (ARS) version of this feature allows users to place calls by
dialing the full public-network numbers without first having to dial a Feature Access Code (FAC),
such as the number “9” to access an outside line. The system recognizes the call as an ARS
call and uses the ARS digit analysis and digit conversion tables to manipulate the digits to route
the call.
The Automatic Alternate Routing (AAR) version of this feature is similar except that the call is
routed as an AAR call and therefore uses the AAR digit analysis and digit conversion tables.
AAR/ARS overlap sending
Communication Manager supports overlap sending for AAR and ARS calls that are routed over
ISDN-PRI trunk groups. ISDN-PRI call-address information is sent one digit at a time instead of
in one block. In countries with complex public-network numbering plans, this allows for a
significant decrease in call setup time. When overlap receiving is enabled, this is especially
significant for tandem calls.
AAR/ARS partitioning
Allows AAR and ARS to be partitioned into 8 user groups within a single system and provides
individual routing treatment for each of these user groups.
User groups share the same Partition Group Number, which indicates the choice of routing
tables that are used on a particular call. Each Class of Restriction (COR) is assigned a specific
Partition Group Number or Time of Day specification. Different classes of restriction may be
assigned the same Partition Group Number.
182 Avaya AuraTM Communication Manager Overview
Enbloc Dialing and Call Type Digit Analysis
AAR/ARS partitioning
Allows AAR and ARS to be partitioned into 8 user groups within a single system and provides
individual routing treatment for each of these user groups.
Enbloc Dialing and Call Type Digit Analysis
The Enbloc Dialing and Call Type Digit Analysis feature allows users to automatically place
outgoing calls based on the telephone number information in the telephone’s call log, without
the user having to modify the telephone number.
Enbloc Dialing is the ability of Communication Manager software to receive in one message all
of the digits necessary to set up a call, or any time the terminal makes a request rather than
receiving digits one at a time.
Call Type Digit Analysis is the ability for Communication Manager to decide how to route a call
based on all of the digits in a telephone number in a call log. The system administrator
maintains the Call Type Digit Analysis Table, which is similar to AAR and ARS Routing Tables.
A phone signals for Call Type Digit Analysis when the user places a call from the phone’s call
log or from the contacts or corporate directory. The telephone sends to Communication
Manager the information stored in the call log, or from the list of contacts or the corporate
directory. Communication Manager uses the Call Type Digit Analysis Table to analyze the
stored number, determines a corresponding dialable number, and makes the call.
The Enbloc Dialing and Call Type Digit Analysis feature is available with Communication
Manager. It works in conjunction with Avaya terminals (proprietary signaling protocol): DCP
phones, H.323 and SIP terminals (96xx H.323 phones, Avaya one-X Mobile Edition for S60 3rd
Edition Dual-Mode SIP phones).
Generalized route selection
Provides voice and data call-routing capabilities. You use it to select not only the least-cost
routing, but also optimal routing over the appropriate facilities. It enhances AAR and ARS by
providing additional parameters in the routing decision and maximizing the chance of using the
right facility to route the call. Also, if an endpoint incompatibility exists, it provides a conversion
resource (such as a modem from a modem pool) to attempt to match the right facility with the
right endpoint.
Issue 6 May 2009
183
Call routing
Look-ahead routing
Provides an efficient way to use trunking facilities. It allows you to continue to try to reroute an
outgoing ISDN-PRI call that is not completing. When Communication Manager receives a
cause value that indicates congestion, Look-Ahead Routing tells the system what to do next.
For each routing preference, you can indicate if the next routing-preference should be
attempted or if the current routing-preference should be attempted again.
Node number routing
Allows you to specify the route pattern associated with each node in a private network. It is a
required capability for Extension Number Portability and is used in conjunction with Automatic
Route Selection, AAR and ARS Partitioning, Private Networking, and Uniform Dial Plan.
Uniform Dial Plan extensions can be routed to a specified node using its associated pattern.
Node Number Routing allows a Uniform Dial Plan route pattern based on node numbers or on
location codes. On the AAR and ARS Digit Analysis Tables, you also can specify a Node
Number instead of a Route Pattern.
Time of day routing
Provides the most economical routing of ARS and AAR calls. This routing is based on the time
of day and day of the week that each call is made. Up to 8 TOD routing plans may be
administered, each scheduled to change up to 6 times a day for each day in the week. This
allows you to take advantage of lower calling rates during specific times of the day and week. In
addition, companies with locations in different time zones can use different locations that have
lower rates at different times of the day or week. This feature is also used to change patterns
during the times an office is closed in order to reduce or eliminate unauthorized calls.
Multiple location support
Multiple Location Support enables local user time, local ARS Public Analysis Tables for local
trunking, automatic Daylight Savings Time, and enhances shared resource algorithms (touch
tone receivers) when Remote Expansion Port Networks (EPNs), ATM Port Networks, and
Avaya Media Gateways are remoted off of a central server at a different location.
184 Avaya AuraTM Communication Manager Overview
Traveling class marks
Multiple location support for network regions
See Multiple location support for network regions on page 151.
Traveling class marks
Traveling Class Marks are a mechanism for passing the facility restriction level of a caller from
one Electronic Tandem Network switch to another. Traveling Class Marks allow privilege
checking to be passed across switches through the Electronic Tandem Network.
Answer detection
For purposes of Call-Detail Recording (CDR), it is important to know when the called party
answers a call. Communication Manager provides three ways to determine whether the called
party has answered an outgoing call.
Answer supervision by time-out
You set a timer for each trunk group. If the caller is off-hook when the timer expires,
Communication Manager assumes that the call has been answered. This is the least accurate
method. Calls that are shorter than the timer duration do not generate call records, and calls
that ring for a long time produce call records whether they are answered or not.
Call-classifier board
A call-classifier board detects tones and voice-frequency signals on the line and determines
whether a call has been answered.
Network answer supervision
The Central Office (CO) sends back a signal to indicate that the far end has answered. If a call
has traveled over a private network before reaching the CO, the signal is transmitted back over
Issue 6 May 2009
185
Call routing
the private network to the originating system. This method is extremely accurate, but is not
available in the United States over CO, FX, or WATS trunks.
186 Avaya AuraTM Communication Manager Overview
Chapter 17: Reliability and survivability
Alternate gatekeeper
The alternate gatekeeper enhancement can provide survivability between Communication
Manager and IP communications devices such as IP Telephones and IP Softphone. This is
accomplished by providing alternate gatekeepers (CLAN) in the event of network or gatekeeper
failure and by load balancing endpoint traffic among multiple gatekeepers. It is important to
recognize that calls will drop during that interval while the communication is re-established to
the switch.
The Alternate Gatekeeper List (AGL) feature allows administrators to specify the number of IP
interfaces for each connected network region that are allowed for phones within a specific
network region.
Auto fallback to primary for H.248 gateways
This feature automatically returns a fragmented network, where a number of H.248 media
gateways are being serviced by one or more Local Survivable Processors (LSP), to the primary
Avaya 8XXX Server. This feature is targeted to H.248 media gateways only.
This feature allows the administrator to define any of the following rules for migration:
l
l
l
l
l
Whether or not the media gateways, serviced by LSPs, should automatically migrate to the
primary media gateway.
Whether or not the media gateway should migrate immediately when possible, regardless
of active call count.
Whether or not the media gateway should only migrate if the active call count is 0.
Whether or not the media gateway should only be allowed to migrate within a window of
opportunity, by providing day of the week and time intervals per day. This option does not
take call count into consideration.
Whether or not the media gateway should be migrated within a window of opportunity by
providing day of the week and time of day, or immediately if the call count reaches 0. Both
rules are active at the same time.
Internally, the primary call controller gives priority to registration requests from those media
gateways that are currently not being serviced by an LSP. This priority is not administrable.
Issue 6 May 2009
187
Reliability and survivability
Connection preserving failover/failback for H.248 media
gateways
The Connection Preserving Migration (CPM) feature preserves existing bearer (voice)
connections while an H.248 media gateway migrates from one Communication Manager server
to another. Migration might be caused by a network or server failure.
Only stable calls are preserved. Call that are not preserved are:
l
Unstable calls. An unstable is any call where the call talk path between parties has not
been established, or is not currently established. Some examples are:
- Calls with dial tone
- Calls in dialing stage
- Calls in ringing stage
- Calls listening to announcements
- Calls listening to music
- Calls on hold (soft, hard)
- Calls in ACD queues
- Calls in vector processing
l
IP trunks, both SIP and H.323
l
ISDN-BRI telephones
l
ISDN-BRI trunks
Users on connection-preserved calls cannot use such features as Hold, Conference, or
Transfer.
Connection preserving upgrades for duplex servers
The connection preserving upgrades for duplex servers feature provides connection
preservation on upgrades of duplex servers for:
l
connections involving IP telephones
l
connections involving TDM connections on port networks
l
connections on H.248 gateways
l
IP connections between port networks and media gateways
Stable calls are preserved. Unstable calls are dropped.
188 Avaya AuraTM Communication Manager Overview
Enterprise Survivable Servers
Enterprise Survivable Servers
The Enterprise Survivable Server (ESS) provide survivability by allowing backup servers to be
placed in various locations in the customer network. The backup servers supply service to port
networks in the case where the Avaya 8XXX Server pair fails, or connectivity to the main server
or server pair is lost.
In an ESS environment, there can only be one main server, either one S85XX Server, or one
pair of S87XX servers. If the main server is an S85XX Server, all ESSs in the configuration must
also be S85XX Servers. During normal operation, the main server communicates with and
controls all the port networks. The main server contains a license file that identifies the server
as the main server and activates the ESS functionality.
The S8400 server can be used as an ESS in Communication Manager Release 5.2 and later.
For more information, see Using Avaya Enterprise Survivable Servers (ESS), 03-300428.
Automatic return to primary server
When ESS is in control due to a network fragmentation or catastrophic main server failure, the
return to the primary (main) server is predicated by three options:
l
scheduled
l
manual
l
automatic
There is a timer that is associated with the auto return to primary feature. The customer sets the
timer before the auto return to primary feature is activated to prevent recovery to the main
server before the network is stable.
Dial Plan Transparency for LSP and ESS
The Dial Plan Transparency feature preserves users’ dialing patterns if a media gateway
registers with a local survivable processor (LSP), or when a port network registers with an
Enterprise Survivable Server (ESS).
When a media gateway registers with a local survivable processor (LSP), or when a port
network registers with an Enterprise Survivable Server (ESS), the Dial Plan Transparency
feature routes calls over the public network when they cannot be routed over the IP network.
Issue 6 May 2009
189
Reliability and survivability
Note:
This feature lets users continue their dialing patterns when LSP or ESS
fragments exist, but it does not guarantee feature transparency for the calls. In
most cases, only basic trunk features are available to the calling and called
parties.
Note:
IP bearer duplication using the TN2602AP circuit pack
The TN2602AP IP Media Resource 320 circuit pack provides high-capacity voice over Internet
protocol (VoIP) audio access to the switch for local stations and outside trunks. The IP Media
Resource 320 provides audio processing for the following types of calls:
l
TDM-to-IP
l
IP-to-TDM
l
IP-to-IP
The TN2602AP IP Media Resource 320 circuit pack has two capacity options, both of which are
determined by the license file installed on Communication Manager:
l
320 voice channels, considered the standard IP Media Resource 320
l
80 voice channels, considered the low-density IP Media Resource 320
Only two TN2602AP circuit packs are allowed per port network.
Note:
Note:
The TN2602AP IP Media Resource 320 is not supported in CMC1 and G600
Media Gateways. For more information about the TN2602AP circuit pack, see
Avaya Aura™ Communication Manager Hardware Description and Reference,
555-245-207.
Load balancing
Up to two TN2602AP circuit packs may be installed in a single port network for load balancing.
The TN2602AP circuit pack is also compatible with and can share load balancing with the
TN2302 and TN802B IP Media Processor circuit packs. Actual capacity may be affected by a
variety of factors, including the codec used for a call and fax support.
190 Avaya AuraTM Communication Manager Overview
IP bearer duplication using the TN2602AP circuit pack
Bearer signal duplication
Two TN2602AP circuit packs may be installed in a single port network for bearer signal
duplication. In this configuration, one TN2602AP is an active IP media processor and one is a
standby IP media processor. If the active media processor, or connections to it, fail, active
connections failover to the standby media processor and remain active. This duplication
prevents active calls in progress from being dropped in case of failure.
For bearer duplication, both TN2602AP circuit packs must be Hardware Version 2, and must
have firmware version 211 or higher.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication
feature of the TN2602AP circuit pack. If these telephones are used while an
interchange from active to standby media processor is in process, calls might be
dropped.
Note:
!
Important:
Important:
If you change from load balanced to duplicated TN2602s, existing calls retain the
real IP address on the TN2602AP circuit pack. New calls are associated with the
virtual IP address of the TN2602AP circuit pack. If an interchange occurs during
this time, existing calls that are associated with the real IP address will drop.
Reduced channels with duplicated TN2602AP circuit packs
If a pair of TN2602AP circuit packs, previously used for load balancing, are re-administered to
be used for bearer duplication, only the voice channels of the active circuit pack can be used.
For example,
l
l
l
If you have two TN2602 AP circuit packs in a load balancing configuration, each with 80
voice channels, and you re-administer the circuit packs to be in bearer duplication mode,
you have 80, not 160, channels available.
If you have two TN2602 AP circuit packs in a load balancing configuration, each with 320
voice channels, and you re-administer the circuit packs to be in bearer duplication mode,
you will have 320, not the maximum 484, channels available.
When two TN2602AP circuit packs, each with 320 voice channels, are used for load
balancing within a port network, the total number of voice channels available is 484, not
640. The reason is that 484 is the maximum number of time slots available for connections
within a port network.
Issue 6 May 2009
191
Reliability and survivability
IP endpoint Time-to-Service
The IP endpoint Time-to-Service (TTS) feature improves a customer’s IP endpoint time to
service, especially in cases where the system has a lot of IP endpoints trying to register or
re-register. With this feature, the system considers that IP endpoints are in-service immediately
after they register.
Local Survivable Processor
A Local Survivable Processor (LSP) is an Internal Call Controller (ICC) with an integral G700
Media Gateway, in which the ICC is administered to behave as a spare processor rather than as
the main processor. The standby Avaya S87XX Server runs in duplex mode with the main
server ready to take control in the event of a outage with no loss of communication.
An LSP is a configuration used to provide redundancy of the Avaya call processing application.
Usually, a media module serves as the ICC for the system, but it can also serve as a redundant
processor for call processing. In the LSP configuration, the processor serves as an alternate
controller/gatekeeper for IP entities, such as IP telephones and media gateways. These IP
entities use the LSP when they lose connectivity to their primary controller.
In the event that the communication link is broken between the remote Avaya G700 Media
Gateway and the primary call controller (either an Avaya S8300 Server or an Avaya S87XX
Server), the LSP provides service for the Avaya IP telephones and Avaya G700 Media
Gateways that were controlled by the primary call controller.
How the Avaya G700 Gateways and IP endpoints change control from the primary to the LSP is
driven by the endpoints themselves, using a list of call controllers. During initialization, each IP
endpoint and Avaya G700 Gateway receives a list of call controllers. The IP endpoints ask each
call controller in the list for service until one responds with a positive reply. If the link to that call
controller fails at some later time, the endpoint will try to receive service from the other call
controllers in the list, including the LSP.
The LSP provides service to all Avaya G700 Gateways and IP endpoints that can sometimes
register with it. When the primary call controller is prepared to provide service, the LSP is reset.
This informs the IP endpoints to try their call controller list again, and returns to the primary call
controller for service.
The LSP provides redundancy in a variety of configurations, and can be located anywhere in a
network of Avaya G700 Gateways.
For LSP capacities, refer to the capacities table. See Capacities on page 26 for instructions how
to view the capacities table.
192 Avaya AuraTM Communication Manager Overview
Handling of split registrations
Note:
You cannot use Upgrade Tool or Avaya Installation Wizard (AIW) for the upgrade
procedures to Communication Manager Release 5.2 or later.
Note:
Handling of split registrations
Split registrations occur when resources on one network region are registered to different
servers. For example, after an outage activates Local Survivable Processors (LSPs),
telephones in a network region register to the main server or Enterprise Survivable Server
(ESS), while the gateways in that network region are registered with an LSP. The telephones
registered with the main server are isolated from their trunk resources. You can manage split
registrations by:
l
l
Setting the Migrate H.248 MG to primary field on the system-parameters
mg-recovery-rule screen to Immediately.
Forcing telephones and gateways to register with the main server or the LSP. Split
registrations occurring between a main server and LSPs or between an ESS and LSPs are
managed by this feature. This feature does not handle split registration between a main
server and an ESS.
Multiple network regions per CLAN
See Multiple network regions per CLAN on page 151.
Power failure transfer
Provides service to and from the local telephone company central office (CO), including wide
area telecommunications system, during a power failure. This allows you to make or answer
important or emergency calls during a power failure. This feature is also called emergency
transfer.
Issue 6 May 2009
193
Reliability and survivability
Standard Local Survivability
Standard Local Survivability (SLS) provides a local Avaya G250/G350 Media Gateway and
Juniper J4350/J6350 gateways with a limited subset of Communication Manager functionality
when there is no IP-routed WAN link available to the main server or when the main server is
unavailable.
SLS provides:
l
l
l
l
l
Calling capability among analog, DCP, and IP telephones
ISDN BRI/PRI trunk interfaces supported on the G250-DS1, G250-BRI, G350, and Juniper
J4350/J6350 gateways
Non-ISDN digital DS1 trunk interfaces supported on the G250-DS1, G350 and Juniper
J4350/J6350 gateways
Outbound dialing through the local PSTN (local trunk gateway) from analog, DCP, and IP
telephones
Inbound calls from each trunk to pre-configured local analog or IP telephones that have
registered
l
Direct Inward Dialing (SLS)
l
Multiple call appearances
l
Hold (SLS) and Call Transfer (SLS) functions
l
Contact closure feature
l
Local call progress tones (dial tone, busy, etc.)
l
Emergency Transfer in survivable mode on the media gateway hardware in cases of
power loss
l
Auto Fallback to Primary Server
l
IP station registration
l
Expanded dial extension numbering for a maximum of 13 digits
Survivable Remote EPN
The Survivable Remote Expansion Port Network (SREPN) allows a DEFINITY ECS (R6r or
later) EPN to provide service to the customer when the link to the main processor fails or is
severed or when the processor or CSS fails. When the links to the system are restored and
stable, the logic switch is manually reset and the EPN is reconnected to the links from the
194 Avaya AuraTM Communication Manager Overview
Survivable Remote EPN
switch. There are both command and manual resets. The resets can be done remotely at the
SAT or manually at the equipment.
The SREPN must be administered separately (not as a duplicated PPN) to function in a disaster
recovery scenario. It does not function as a survivable remote EPN without the administration
(stations, trunks, features) to support its operation.
Note:
Note:
SREPN is not compatible with ATM port network connectivity (ATM-PNC).
Issue 6 May 2009
195
Reliability and survivability
196 Avaya AuraTM Communication Manager Overview
Chapter 18: Security, privacy, and safety
System administrator
Authentication, Authorization, and Accounting Services
Authentication, Authorization and Accounting (AAA) Services allow customers to store and
maintain administrator account information on a central server. Communication Manager
supports account information being stored on an external AAA server or locally on the
Communication Manager server itself. Both types of accounts may be used at the same time.
AAA Service interactions with Communication Manager use the Pluggable Authentication
Module (PAM) and Name-Switch Service (NSS) features of Linux, which are provided on
Linux-based Communication Manager servers.
External AAA support is a Linux process that is separate from Communication Manager, is not
controlled by a license file, and is freely available to the customer. Customers can use the same
AAA server for Communication Manager as is used by other servers on their network.
Access security gateway
Access security gateway (ASG) is an authentication interface used to secure the system
administration and maintenance ports and/or logins on the system. Access security gateway
employs a challenge/response protocol to confirm the validity of a user and reduce the
opportunity for unauthorized access.
Successful authentication is accomplished when the feature communicates with a compatible
key. The challenge/response negotiation is initiated once an RS-232 session is established and
a valid system login ID has been supplied by a user. The authentication transaction consists of
a challenge, issued by the system and based on the login ID supplied by the user, followed by
receipt of the expected response, which is supplied by the user.
Branch gateway enhancements
Access security gateway authentication allows Avaya’s Services organization to remotely login
to Avaya 8XXX Servers that are under service agreement. To enhance customer support in
branch offices, ASG is enhanced to allow authentication capabilities to the G350/G250 gateway
product line.
Issue 6 May 2009
197
Security, privacy, and safety
Alternate facility restriction levels
This feature allows Communication Manager to adjust facility restriction levels or authorization
codes for lines or trunks. Each line or trunk is normally assigned a facility restriction level. With
this feature, alternate facility restriction levels are also assigned. Attendants can change to the
alternates, thus changing access to lines and trunks.
You might want to use this feature to disable most long-distance calling at night, for example, to
prevent unauthorized staff from making long-distance calls.
! CAUTION:
This feature may change the AAR and ARS routing preferences. Using it on
tandem and tie-trunk applications affects entire networks. Calls that are part of a
cross-country private network may be blocked.
CAUTION:
Alternate operations support system alarm number
This feature allows you to establish a second number for Communication Manager to call when
an alarm event occurs. This feature is useful for alerting a second support organization, such as
INADS or OneVision.
Privacy - attendant lockout
See Attendant lockout - privacy on page 39.
Authorization codes - 13 digits
Authorization codes extend calling-privilege control and enhance security for remote-access
callers. Authorization codes can be up to 13 digits in length.
Avaya site administration authorization codes may be used to:
l
l
Override facility restriction levels assigned to originating stations or trunks
Restrict individual incoming tie trunks and remote-access trunks from accessing outgoing
trunks
l
Track CDR calls for cost-allocation purposes
l
Provide additional security control
198 Avaya AuraTM Communication Manager Overview
System administrator
Call restrictions
By dialing an access code, administrators and attendants have the ability to restrict users from
making or receiving certain types of calls. There are five restrictions:
l
Outward. User cannot place external calls.
l
Station-to-station. User cannot place or receive internal calls.
l
Termination. User cannot receive any calls (except priority calls).
l
Toll. User cannot place toll calls but can place local calls.
l
Total. User can neither place nor receive any calls.
Class of Restriction
Defines many different classes of call origination and termination privileges. Communication
Manager may have no restrictions, only a single COR, or may have as many classes of
restrictions as necessary to effect the desired restrictions. Many different types of classes of
restriction can be assigned to many types of facilities on the switch. For example, you can use a
calling-party COR to prevent callers from accessing the public network.
Block collect call
See Block collect call on page 96.
Customer-provided equipment alarm
Provides you with an indication that a system alarm has occurred and that the system has
attempted to contact a service organization. A device that you provide, such a lamp or a bell, is
used to indicate the alarm situation. You can administer the level of alarm about which you want
to be notified.
Data privacy
See Data privacy on page 175.
Issue 6 May 2009
199
Security, privacy, and safety
Data restriction
See Data restriction on page 176.
Encryption algorithm for bearer channels
Communication Manager supports the Advanced Encryption Standard (AES) format of signal
encryption for IP telephony. This encryption algorithm is in addition to the Avaya proprietary
encryption protocol, the Avaya Encryption Algorithm (AEA).
AES encryption is a cryptographic algorithm developed by the U.S. Government to protect
unclassified information. Communication Manager uses AES with 128 bit keys in counter mode
(AES-128-CTR).
Administration is supported to select a combination of no encryption, AEA encryption, and/or
AES encryption on a per codec set basis.
SRTP media encryption
SRTP is a media encryption standard. An end-to-end SRTP implementation includes the
following media processing platforms:
Note:
l
SIP telephones (SRTP video encryption for SIP is not covered)
l
TN2602AP circuit pack
l
MM760 media module in a G700 Media Gateway
l
G700 VoIP on-board element
l
G450 VoIP element
l
G350 VoIP element
l
G250 VoIP element
Note:
All of these platforms, except for SIP telephones, also support AES and AEA
media encryption.
200 Avaya AuraTM Communication Manager Overview
System administrator
Enhanced security logging
Enhanced security logging increases the granularity of logging of user activity, and provides a
single place of customer choosing, external server or Linux syslog, to store the security logs.
Along with centralized authentication, enhanced security logging consolidates several existing
Communication Manager security log files, and routes the files to an industry standard external
log server or the internal Linux syslog.
Enhanced security logging adds these additional security events:
Note:
l
Successful and failed admin login
l
Logout
l
Successful and failed endpoint authentication
l
DOS attacks
l
SAT administration changes, including data describing the exact change
Note:
To take advantage of centralized authentication, the customer must have an
industry standard RADIUS, Microsoft ActiveDirectory, or LDAP external server.
These are not supplied by Avaya.
Facility restriction levels and traveling class marks
Allows certain calls to specific users, while denying the same calls to other users. For example,
certain users may be allowed to use Central Office (CO) trunks to other corporate locations
while other users may be restricted to less expensive private-network lines. You can
administer up to eight levels of restriction for users of AAR and ARS.
H.248 link encryption
To provide privacy for media streams carried over IP networks, the H.248 signaling channel
between the Avaya 8XXX Server (media gateway controller) and the media gateways is
encrypted. This signaling channel is used to distribute the media session keys to the media
gateways, and may carry user-dialed authorization codes and passwords.
This feature protects our customer investments by encrypting the signaling channel between
the H.248 gateway and server. This feature also protects the media encryption key, PINs, and
account codes between the media gateway and the media gateway controller.
Encryption of the H.248 link to any given media gateway may be enabled or disabled through
the Media Gateway screen. However, the encryption protocol cannot be disabled.
Issue 6 May 2009
201
Security, privacy, and safety
Malicious call trace
Allows you to trace malicious calls. You define a group of terminal users who can notify others in
the group when they receive a malicious call. These users can then retrieve information related
to the call. Using this information, you can identify the malicious call source or provide
information to personnel at an adjacent system to complete the trace. It also allows you to
record the malicious call, as well as trace a malicious call over ETSI PRI.
Malicious call trace logging
Malicious call trace logging allows a PC to receive information from Communication Manager to
log malicious calls.
Mask station name and number for internal calls
Communication Manager blocks the calling party name and number for internal calls. This
feature includes the following capabilities:
l
l
l
l
Called party does not see the name or number of the calling party.
Display of the incoming call is administrable on a system level. For example, customers
may administer the display to show “Restricted Call.”
Blocking of the calling party name and number is administrable on a Class of Restriction
(COR) basis, as well as on a call basis (through the CPN Block button).
The calling party name and number information is available to CDR, vectors, and/or AE
Services.
Media encryption
Media Encryption is the encryption of the audio (voice) portion of a Voice Over IP (VoIP) call.
Media Encryption can be used to provide enhanced privacy for VoIP communications that
involve exchange of sensitive information. Media Encryption is provided between Avaya media
gateways and Avaya 8XXX Servers.
Digitally encrypting the audio (voice) portion of a VoIP call can reduce the risk of electronic
eavesdropping. IP packet monitors, sometimes called sniffers, are to VoIP calls what wiretaps
are to circuit-switched (TDM) calls. One exception is that an IP packet monitor can watch for
and capture unencrypted IP packets, and can play back the conversation in real-time or store it
for later playback.
202 Avaya AuraTM Communication Manager Overview
System administrator
Communication Manager encrypts IP packets before they traverse the IP network. An
encrypted conversation sounds like white noise or static when played through an IP monitor.
End users do not know that a call is encrypted because there are:
l
No visual or audible indicators to indicate that the call is encrypted.
l
No appreciable voice quality differences between encrypted calls and non-encrypted calls.
!
SECURITY ALERT:
SECURITY ALERT:
Be sure that you understand these important media encryption limitations:
- Any call that involves a circuit-switched (TDM) endpoint, such as a DCP or analog
telephone, is vulnerable to conventional wire-tapping techniques.
- Any call that involves an IP endpoint or gateway that does not support encryption
can be a potential target for IP monitoring. A common example of this is are
IP trunks to 3rd-party vendor switches.
- Any party that is not encrypting an IP conference call exposes all parties on the IP
call between the unencrypted party and its supporting media processor to
monitoring, even though the other IP links are encrypting.
For a list of the supported hardware, software, and firmware requirements for Media Encryption,
and a list of the equipment that is not supported, see Avaya Aura™ Communication Manager
Hardware Description and Reference, 555-245-207.
License file requirements
Media Encryption does not work unless the server has a valid License File with Media
Encryption enabled. To determine whether Media Encryption is enabled in the current License
File:
1. Type display system-parameters customer-options a. Press Enter.
The system displays the Optional Features screen.
2. Press Next until you see the Media Encryption Over IP field (Figure 6: Optional Features
screen on page 204).
3. Ensure that the Media Encryption Over IP field is set to y.
Issue 6 May 2009
203
Security, privacy, and safety
Figure 6: Optional Features screen
display system-parameters customer-options
OPTIONAL FEATURES
Emergency Access to Attendant?
Enable 'dadmin' Login?
Enhanced Conferencing?
Enhanced EC500?
Enterprise Survivable Server?
Enterprise Wide Licensing?
ESS Administration?
Extended Cvg/Fwd Admin?
External Device Alarm Admin?
Five Port Networks Max Per MCC?
Flexible Billing?
Forced Entry of Account Codes?
Global Call Classification?
Hospitality (Basic)?
Hospitality (G3V3 Enhancements)?
IP Trunks?
y
y
y
y
n
n
n
y
n
n
n
n
n
y
n
y
Page
4 of
11
IP Stations? y
ISDN Feature Plus?
ISDN/SIP Network Call Redirection?
ISDN-BRI Trunks?
ISDN-PRI?
Local Survivable Processor?
Malicious Call Trace?
Media Encryption Over IP?
Mode Code for Centralized Voice Mail?
n
n
y
y
n
n
y
n
Multifrequency Signaling?
Multimedia Call Handling (Basic)?
Multimedia Call Handling (Enhanced)?
Multimedia IP SIP Trunking?
y
y
y
y
IP Attendant Consoles? y
(NOTE: You must logoff & login to effect the permission changes.)
PIN Checking for Private Calls
This feature restricts users from making private calls (internal or external) by forcing them to
enter a Personal Identification Number (PIN) code after dialing a PIN feature access code and
only when the PIN is valid, the user can dial the destination digits to make a call. The PIN code
used for the call is reported in Call Detail Record (CDR) output with a special character P. SIP
telephones do not support PIN checking.
Restriction - controlled
Allows an attendant or telephone user, with console permission, to activate and deactivate for
an individual telephone or a group of telephones, the following restrictions:
l
outward
l
total
l
station-to-station
l
termination restrictions
204 Avaya AuraTM Communication Manager Overview
System administrator
Secure shell and secure FTP
The Telnet protocol allows remote access to a network device console that is based on login
and password authentication. Beginning with Communication Manager release 3.0, Secure
Shell (SSH) provides this capability over an encrypted channel. Similarly, Secure FTP (SFTP) is
an encrypted version of the FTP protocol that allows remote file transfers. SSH/SFTP provides
a secure alternative for file transfer of firmware download files and voice announcements, as
well as secure remote server access.
The enable filexfer command enables SSH and SFTP for both the TN799DP Control LAN
(CLAN) circuit pack, and the TN2501AP Voice Announcement over LAN (VAL) circuit pack.
The Telnet, FTP, SSH, and SFTP enabling capabilities on the TN2312A/BP IP Server Interface
(IPSI) circuit pack continue to be handled through the Communication Manager System
Management Interface and the Communication Manager Linux bash shell.
Security of IP telephone config files
This feature supports the inclusion of a digital certificate and the use of TLS to allow an IP
telephone to authenticate the server for the download of configuration files. This enables IP
telephones to ensure that configuration parameters come only from an authenticated source.
Configuration files that are delivered through this mechanism can deliver message digest
values for the authentication of software code files delivered through a non-secure connection.
Security of IP telephone registration/H.323 signaling channel
Note:
Please check with your Avaya Sales Representative or your Avaya Authorized
Business Partner for availability of this feature.
Note:
This feature provides a secure mechanism for an H.323 endpoint and a Communication
Manager gatekeeper to mutually authenticate themselves and the contents of the messages
that they exchange during IP registration, admission, and status (RAS). This authentication is
based on the simple 3-to-8 digit PIN administered for each extension.
Execution of Encrypted Key Exchange (EKE) procedures during RAS results in the negotiation
of a strong secret that is shared between the endpoint and the gatekeeper. This strong secret is
used to derive a set of secrets which are used to digitally sign all RAS and call signaling
messages, and to encrypt selected elements of call signaling messages, media session keys
and CCMS messages. If one or the other parties does not possess the correct PIN, the
computed shared secrets will, in fact, be different. Message authentication fails, and the parties
refuse to communicate.
In summary, these procedures permit:
l
The endpoint and the gatekeeper to authenticate each other;
Issue 6 May 2009
205
Security, privacy, and safety
l
l
l
l
l
The endpoint and the gatekeeper to sign/authenticate each message sent;
Privacy for selected elements of call signaling, including media session encryption keys
and dialed digits.
Security of the endpoint/gatekeeper communication even if an observer obtains the user's
PIN.
Security of past or future communications even if one session is penetrated by an attacker
with knowledge of the PIN. (This is known as “perfect forward secrecy”.)
Reuse of the negotiated strong secret (identified by a unique session ID) to secure new
signaling links between parties for re-registration or trunking.
Security Violation Notification
Security Violation Notification (SVN) allows you to set security-related parameters and to
receive notification when the limits that you have established are violated. You can run reports
related to both valid and invalid access attempts. You can also disable a login ID or remote
access authorization that is associated with a security violation.
Signaling encryption for SIP trunks
Signaling encryption for SIP trunks protects customer investments by encrypting the voice
channel over SIP trunks. Signaling encryption for SIP trunks is accomplished by using TLS and
protects customer investments by encrypting signaling data and instant messages.
Station security codes
To provide additional security around the customer options the “init” login has been provided
with additional security for the purpose of establishing an authentication procedure for attempts
to remotely log into the system.
Tripwire security
Tripwire is a security program provided on all Linux-based Avaya 8XXX Servers. The list of files
that Tripwire monitors needs to be determined during design when all administration and
configuration files have been identified.
If there are any detected security violations, Tripwire reports its findings through the security log.
These events generate an alarm.
206 Avaya AuraTM Communication Manager Overview
End user
Note:
Note:
Tripwire normally reports security violations through e-mail. However, by
reporting events to the security log, security violations can be immediately acted
upon.
End user
Backup alerting
Notifies backup attendants that the primary attendant cannot pick up a call. It provides both
audible and visual alerting to backup stations when the attendant queue reaches its queue
warning level. When the queue drops below the queue warning level, alerting stops. Audible
alerting also occurs when the attendant console is in night mode, regardless of the attendant
queue size.
Barrier codes
A barrier code is a security code that is used with remote access to prevent unauthorized
access to your system. To increase your system security, use a 7-digit barrier code with remote
access barrier code aging. A barrier code automatically expires if an expiration date or number
of accesses has exceeded the limits you set. If both a time interval and access limits are
administered for a barrier code, the barrier code expires when one of the conditions is satisfied.
Note:
Note:
Barrier codes are not tracked by call detail recording (CDR). Barrier codes are
incoming access codes, whereas, authorization codes are primarily outgoing
access codes.
Calling/Connected Party Number restriction
Per call CPN restriction
Users may indicate calling number privacy information. For ISDN calls, the CPN presentation
indicator is encoded accordingly. For non-ISDN calls going to a public network that supports the
CPN restriction feature, the network specific feature activation code gets passed to the network
for interpretation and activation of the desired feature.
Issue 6 May 2009
207
Security, privacy, and safety
If per call CPN restriction is activated for an outgoing call, it will override any per line CPN
restriction administration for the calling station, and will override any ISDN trunk group
administration for sending calling number.
Per line CPN restriction
Users may block the calling party number when originating calls. For ISDN calls, the CPN
presentation indicator is encoded accordingly. For non-ISDN calls, going to a public network
that supports the CPN restriction feature, the network specific feature activation code gets
passed to the network for interpretation and activation.
If per line CPN restriction is administered for a station, it will override any ISDN trunk group
administration for sending calling party number.
Crisis alerts to a digital numeric pager
Crisis alert can also send notification of an emergency call to a digital pager. In this case, it
sends a message of 7-digits to 22-digits to the pager and displays a crisis alert code, an
extension and room number, and a main number (if one is entered). The person paged thus
knows the origin of the emergency call and can direct emergency-service response to the
appropriate location.
To use crisis alert with a digital pager, the system is administered so that at least one digital set
has a CRSS-ALRT button and the Alert Pager field is set to y. Any station with a
CRSS-ALRT button and a pager receives the correct alert.
Crisis alerts to a digital station
Crisis alert uses both audible and visual alerting to notify administered digital display stations
when an emergency call is made. Audible alerting sounds like an ambulance siren. Visual
alerting flashes the CRSS-ALRT button lamp and displays the name and extension, or room, of
the caller. The crisis alert display of the origin of the emergency call enables the attendant or
other user to direct emergency-service response to the caller.
When crisis alerting is active, the station is placed in position-busy mode so that other incoming
calls can not interfere with the emergency call notification. The station can still originate calls to
allow notification of other personnel.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the order
in which they were made. If the system is administered so that only one user must respond, the
first crisis alert remains active at the telephone where it was acknowledged. Subsequent calls
are queued to the next available station in the order in which they were made.
208 Avaya AuraTM Communication Manager Overview
End user
Crisis alerts to an attendant console
Crisis alert uses both audible and visual alerting to notify attendant consoles when an
emergency call is made. Audible alerting sounds like an ambulance siren. Visual alerting
flashes the CRSS-ALRT button lamp and displays the name and extension, or room, of the
caller. The crisis alert display of the origin of the emergency call enables the attendant or other
user to direct emergency-service response to the caller. Though often used in the hospitality
industry, it can be set up to work with any standard attendant console.
When crisis alerting is active, the console is placed in position-busy mode so that other
incoming calls can not interfere with the emergency call notification. The console can still
originate calls to allow notification of other personnel. Once a crisis alert call has arrived at a
console, the console user must press the position-busy button to unbusy the console, and press
the crisis-alert button to deactivate audible and visual alerting.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the order
in which they were made. If the system is administered so that only one user must respond, the
first crisis alert remains active at the telephone where it was acknowledged. Subsequent calls
are queued to the next available station in the order in which they were made.
Emergency access to the attendant
Provides for emergency calls to be placed to an attendant. These calls can be placed
automatically by the system or can be dialed by system users. Emergency access calls can
receive priority handling by the attendant.
E911 CAMA trunk group
See E911 CAMA trunk group on page 134.
Hot Desking Enhancement
Hot Desking is a generic term for features that allow end users to lock and unlock their
telephones or to move fully customized station profiles to another compatible telephone.
Hot Desking enhances the existing features:
l
IP Login/Logoff
l
PSA Association/Dissociation
l
Station Lock and Time of Day Station Lock
Issue 6 May 2009
209
Security, privacy, and safety
Parts of the enhancement require firmware changes for the telephones. Only the 96xx-series
H.323 IP telephones with the appropriate firmware change support the full range of HDE.
Privacy - auto exclusion
When the class of service (COS) is set for the automatic exclusion option, the feature is
activated when you take your telephone off-hook. The feature can be deactivated when you
push the exclusion button before dialing a call or during a call. An excluded call that is on hold
can be taken off hold by any telephone that has a bridged appearance of the telephone that put
the call on hold.
Privacy - manual exclusion
Allows multi-appearance telephone users to keep other users with appearances of the same
extension number from bridging onto an existing call. Exclusion is activated by pressing the
exclusion button on a per-call basis.
Restriction - controlled
See Restriction - controlled on page 204.
Station lock
Station lock allows users to lock their telephones to prevent unauthorized outgoing calls. Users
can block outgoing calls and still receive incoming calls. This feature is activated by pressing a
telephone button or dialing a feature access code (FAC), along with a station security code
(SSC).
Station lock allows users to block all outgoing calls, except for emergency calls, on all
telephones, unless the telephone is pre-administered. An example of a pre-administered
telephone is a telephone that is administered to block all outgoing calls except for emergency
calls. Telephones can be remotely locked and unlocked.
While the Hot Desking Enhancement (HDE) feature is active and a station is locked, the
following restrictions apply:
l
No access to telephone capabilities (applies to 96xx H.323 IP telephones with firmware
changes)
l
No access to Call log
l
No access to Avaya menu
210 Avaya AuraTM Communication Manager Overview
End user
l
No access to Contact list
l
No access to USB
l
No access to Redial button
l
No bridging on EC500 calls
l
No access to bridged appearances
Additionally, if Hot Desking Enhancement is enabled, some telephone capabilities and some
Communication Manager capabilities are locked.
Station lock by Time of Day
The Time of Day (TOD) Station Lock feature allows a customer to lock and unlock one or more
stations using a TOD schedule.
For more information, see Time of day on page 240.
Note:
Note:
Station Lock by TOD cannot override a station that has been manually locked.
The Station Lock by TOD feature cannot release the manually locked station
when the schedule allows the release. If a user unlocks the station that was
locked by the TOD Station Lock feature, the station stays unlocked until the next
TOD lock interval.
Issue 6 May 2009
211
Security, privacy, and safety
212 Avaya AuraTM Communication Manager Overview
Chapter 19: System management
Communication Manager system management provides the administrator powerful tools to
maintain their communication solutions and to drive down the total cost of ownership.
Administration change notification
Administration change notification enables Communication Manager to communicate with the
Avaya Directory Enabled Management (DEM) client. This feature enables the client to have
real-time, integrated, directory-based, read/write access to Communication Manager
administration data based on rules defined by the customer. Administration change notification
enables the client to subscribe to notifications of changes to administration data in
Communication Manager. It thus provides real-time updates whenever administration changes
occur in a particular object (for example, a station).
Administration Without Hardware
See Administration Without Hardware on page 113.
Alternate facility restriction levels
This feature allows Communication Manager to adjust facility restriction levels or authorization
codes for lines or trunks. Each line or trunk is normally assigned a facility restriction level. With
this feature, alternate facility restriction levels are also assigned. Attendants can change to the
alternates, thus changing access to lines and trunks. You might want to use this feature to
disable most long-distance calling at night, for example, to prevent unauthorized staff from
making long-distance calls.
! CAUTION:
CAUTION:
This feature may change the AAR and ARS routing preferences. Using it on
tandem and tie-trunk applications affects entire networks. Calls that are part of a
cross-country private network may be blocked.
Issue 6 May 2009
213
System management
Announcements
Use the Announcements feature to administer announcements that play for callers to your
business. For example, you can inform callers that the call cannot be completed as dialed, the
call is in a queue, or that all lines are busy. An announcement is often used in conjunction with
music. Announcements can be integrated or external.
l
l
Integrated announcements reside on a circuit pack in the carrier.
External announcements are stored on an adjunct, and are played back from the adjunct
equipment.
Authorization codes - 13 digits
See Authorization codes - 13 digits on page 198.
Automatic circuit assurance
Assists in identifying possible trunk problems. Communication Manager maintains a record of
the performance of individual trunks and automatically calls a designated user when a possible
failure is detected. This feature provides better service through early detection of faulty trunks
and consequently reduces out-of-service time.
Automatic transmission measurement system
Measures voice and data trunk facilities for satisfactory transmission performance. The
measurement report contains data on trunk signal loss, noise, signaling return loss, and echo
return loss. Acceptable performance, the scheduling of tests, and report contents are
administrable.
214 Avaya AuraTM Communication Manager Overview
Avaya Directory Enabled Management
Avaya Directory Enabled Management
Avaya Directory Enabled Management (DEM) provides real-time, integrated, directory-based
read/write access to Avaya 8XXX Servers and INTUITY AUDIX messaging servers. It
streamlines workflow and information management in an electronic environment using
converged networks.
DEM creates a meta-directory for converged voice and data networks. It synchronizes directory
information with data from Communication Manager and INTUITY devices, and stores the
information in an LDAP-compliant directory service, for example, eDirectory from Novell, or
active directory from Microsoft. Directory-enabled applications can then use the DEM to
implement workflow processes that automate various system management functions and speed
business operations.
Avaya Integrated Management
Avaya Integrated Management is a systems management software suite that contains
applications to manage a converged voice and data network. The applications include:
l
network management
l
fault management
l
performance management
l
configuration management
l
directory management
l
policy management functionality
Communication Manager configuration manager
Avaya configuration manager provides centralized management of distributed network and
campus environments, using a single point of entry and graphical Web-based interface for
configuration and administration of multiple Avaya 8XXX Servers.
Issue 6 May 2009
215
System management
Communication Manager fault/performance manager
Communication Manager fault/performance manager integrates with Avaya multiservice
network manager to provide a system view of your converged network. Fault manager displays
a hierarchical view of devices and their status, allowing you to view and isolate alarms and
errors. Performance manager provides a comprehensive set of performance reports for
trending and isolation of performance issues.
Avaya site administration
Avaya site administration is a Microsoft Windows-based graphical user interface for making
changes, adding or moving users, and performing basic traffic analysis.
Voice Announcement over LAN manager
Avaya Voice Announcement over LAN (VAL) Manager is part of the Avaya Integrated
Management suite of products. It enables you to the use of a LAN to transfer recorded
announcements to Avaya 8XXX Servers.
Announcements can be stored in .wav files, which can be sent to a voice announcement over
LAN board without conversion. The voice announcement over LAN manager also provides a
repository to backup and restore announcement files, and simplifies administration. With voice
announcement over LAN manager, you can view the current status of announcements, easily
add, change, and remove announcements, and copy and backup announcement files from
Avaya 8XXX Servers to the voice announcement over LAN manager and back, through the
LAN.
Increased announcement support
Communication Manager release 4.0 increases support for announcements, but only on the XL
systems, for example, S8720XL servers. A new license file is required to increase the number
of VAL boards in a customer’s configuration, even if a customer upgrades to an S8720XL.
l
l
l
Support for TN2501 (VAL) boards within a single Communication Manager configuration is
increased from 10 to 128.
The number of touch tone receivers (TTRs) in a system is increased from 1,200 to 8,000.
This does not apply to H.248 gateways since H.248 gateways use a separate TTR
resource.
The total number of supported announcements is increased from 3,000 to 9,000. This
applies to VAL boards (TN2501) and the vVALs. As a result of this increase, the way in
which announcements are administered is also modified.
216 Avaya AuraTM Communication Manager Overview
Barrier codes
l
Note:
The number of announcement files that are supported on the TN2501 circuit pack is
increased from 256 to 1024. This increase is supported on non-XL systems.
Note:
While the TN750 circuit packs still exist, they are no longer supported. In other
words, full functionality is not available on LINUX systems. If used, they are
included in the 128 total announcement boards in the system.
In Communication Manager Release 5.2, the G450 H.248 gateway allocates announcement
time slots dynamically as needed. The Avaya G450 introduces a removable external compact
flash that supports the backup and restore of 1024 announcement files (up to 4 hours of
storage). This requires a memory upgrade to 512MB from the base 256MB.
Avaya VoIP Monitoring Manager
Avaya VoIP monitoring manager (VMON) provides the ability to monitor voice over IP (VoIP)
network quality. This web-based application receives QoS statistics from Avaya IP end points
and displays the data via graphs and reports, so administrators can isolate voice quality
problems and send traps when poor voice quality is detected.
Lightweight Directory Access Protocol
Lightweight Directory Access Protocol version 3 (LDAPv3) is an industry compliant protocol for
accessing online directory services. A directory is like a database, but tends to contain more
description information. Communication Manager integrates with LDAP datastores through the
use of the administration change notification feature and Avaya directory enabled management
client application to provide real-time, integrated, directory-based read/write access to
Communication Manager and INTUITY AUDIX messaging servers.
Barrier codes
See Barrier codes on page 207.
Bulletin board
Provides a place on the switch where you can post information and receive messages from
other switch users, including Avaya personnel. Anyone with appropriate permissions can use
Issue 6 May 2009
217
System management
the bulletin board for everyday messages. In addition, Avaya personnel can leave high-priority
messages that are displayed on the first ten lines of the bulletin board.
Busy verification of telephones and trunks
Allows attendants and users of multi-appearance telephones to make test calls to trunks,
telephones, and hunt groups to check the status of an apparently busy resource. With this
feature, an attendant or multifunction telephone user can distinguish between a telephone that
is truly busy and one that only appears busy because of some problem. You can also use the
feature to quickly identify faulty trunks.
Call charge information
Provides two ways to know the approximate charge for calls made on outgoing trunks:
l
Advice of Charge, for ISDN trunks
Advice of Charge (AOC) collects charge information from the public network for each
outgoing call. Charge advice is a number representing the cost of a call; it is recorded as
either a charging or currency unit.
l
Periodic pulse metering, for non-ISDN trunks
Periodic Pulse Metering (PPM) accumulates pulses transmitted from the public network at
periodic intervals during an outgoing trunk call. At the end of the call, the number of pulses
collected is the basis for determining charges.
Call-charge information helps you to account for the cost of outgoing calls without waiting for the
next bill from your network provider. This is especially important in countries where telephone
bills are not itemized. You can also use this information to let employees know the cost of their
telephone calls, and so encourage them to help manage your company telecommunications
expenses.
Note:
Note:
This feature is not offered by the public network in some countries, including the
United States.
In addition, the pass advice of charge to BRI endpoints feature will transparently pass AOC
information that has been received from PRI networks to WCBRI endpoints.
218 Avaya AuraTM Communication Manager Overview
Call Detail Recording
Call Detail Recording
Records detailed call information on incoming and outgoing calls for the purpose of call
accounting, and sends this call information to a Call Detail Recording (CDR) output device. You
can specify the trunk groups and extensions for which you want records to be kept as well as
the type of information to be recorded. You can keep track of both internal and external calls.
This application contains a wide variety of administrable options and capabilities.
Call Detail Recording display of physical extension
For Expert Agent Selection (EAS) agent-originated calls, if the Record Agent ID on
Outgoing? field on the CDR System Parameters screen is set to y (the default value), then
the agent ID is used for outgoing calls.
If the Record Agent ID on Outgoing? field on the CDR System Parameters screen is set
to n, the physical extension is used.
Legacy CDR and Survivable CDR
You can have Legacy, or Conventional CDR or Survivable CDR for your system. Both methods
provide the identical call accounting information and support the same CDR formats.
In a Legacy CDR environment, Communication Manager generates all CDR records to the
active server and then exports the records to a CDR adjunct (using IP links) for further
processing. In these systems, the CDR adjunct functions in a “listen only” mode receiving the
records sent by Communication Manager. This type of CDR processing was used exclusively in
Communication Manager up through Release 3.X. In a Legacy CDR environment, the system
cannot collect CDR records when link between main, LSP and/or ESS and CDR adjunct is
down (this is generally the case when a LSP or ESS is active). It also requires the IP link
between Communication Manager and the CDR adjunct to be up at all times.
Communication Manager introduced a new method of data collection called Survivable CDR.
Survivable CDR allows CDR data records to be stored on the local hard drive of Communication
Manager, if desired. The CDR adjunct then periodically polls each the Communication Manager
server and collects the CDR data files from the servers. The Survivable CDR is supported on all
Communication Manager platforms, that is, on the main, LSPs, and ESS. Survivable CDR
allows CDR data to be transferred in an encrypted manner from the Communication Manager
server to the CDR adjunct using the Secure File Transfer Protocol (SFTP). Note that Survivable
CDR requires updated CDR adjuncts. CDR records are stored and transferred in batches,
rather than one record at a time as with Legacy CDR.
Issue 6 May 2009
219
System management
Call restrictions
By dialing an access code, administrators and attendants have the ability to restrict users from
making or receiving certain types of calls. There are five restrictions:
l
Outward. The user cannot place external calls.
l
Station-to-station. The user cannot place or receive internal calls.
l
Termination. The user cannot receive any calls (except priority calls).
l
Toll. The user cannot place toll calls but can place local calls.
l
Total. The user can neither place nor receive any calls.
Calling party/billing number
Allows the system to transmit calling party number/billing number (CPN/BN) information to an
ISDN-PRI trunk group. The CPN is the calling party telephone number. BN is the calling party
billing number. The CPN/BN may contain international country codes. It is used with an adjunct
application.
Class of Restriction
See Class of Restriction on page 199.
Class of Service
Defines whether or not telephone users have permission to access features and functions. You
can administer a station to have access to up to 16 Classes of Service. In Communication
Manager, you can also administer Classes of Service on a per-tenant basis if you use
partitioning in your system. You can administer up to 100 COS groups, each with 16 Classes of
Service. This can be useful in controlling service to the stations and attendant of different
tenants.
Examples of features and functions controlled by Class of Service:
l
Automatic callback
220 Avaya AuraTM Communication Manager Overview
Classless Interdomain Routing
l
Call forwarding
l
Data privacy
l
Priority calling
l
Restrict call forwarding off-net
l
Call forward busy/do not answer
l
Extended forwarding and busy/do not answer
l
Personal station access
l
Trunk-to-trunk transfer restriction override
l
Off-hook alert
l
Console permission
l
Client room
Classless Interdomain Routing
See Classless Interdomain Routing on page 151.
Issue 6 May 2009
221
System management
Concurrent user sessions
In order to increase the efficiency of administration and maintenance functions, the
Communication Manager accommodates multiple concurrent administration and maintenance
user sessions. Three or more devices (management terminals or operation support systems)
can be connected to the switch to perform administration and/or maintenance tasks
simultaneously.
Communication Manager supports eight concurrent administration and maintenance users.
Five can perform concurrent administration, and three can perform concurrent maintenance.
The eight concurrent sessions can be in any combination of local and remote connections.
Customer-provided equipment alarm
See Customer-provided equipment alarm on page 199.
Customer telephone activation
Enables customers to install their own telephones, eliminating the need for a service technician
to do the installation. This feature is based on the TTI feature and allows the customer to
associate a physical telephone with a station translations switch.
CTA is a streamlined version of TTI; it has a fixed feature-access code but does not require a
security code. In addition, CTA allows only for “merging” of telephones with station translations,
whereas TTI allows for both “merging” and “unmerging” of telephones with station translations.
CTA applies only to DCP and analog touch-tone, circuit-switched telephones.
DCS automatic circuit assurance
Allows a user or attendant at one node to activate or deactivate automatic circuit assurance
referral calls for the entire DCS network. This transparency allows the referral calls to originate
at a node other than the node that detects the problem.
222 Avaya AuraTM Communication Manager Overview
External device alarming
External device alarming
Allows you to assign analog ports to alarm interfaces for external devices. You can specify a
port location, information to identify the external device, and the alarm level to report when a
contact closure occurs.
Facility busy indication
Allows users of multi-appearance telephones to see which lines, trunk groups, terminating
extension groups, hunt groups, or paging zones (called resources or facilities) are busy. When
the lamp associated with the resource is lit, the resource is busy.
You can store extension numbers, trunk group access codes, and loudspeaker paging access
codes in a facility busy indication button. The facility busy indication button provides direct
access to any of the facilities.
Facility restriction levels and traveling class marks
Allows certain calls to specific users, while denying the same calls to other users. For example,
certain users may be allowed to use central office (CO) trunks to other corporate locations while
other users may be restricted to less expensive private-network lines. You can administer up
to eight levels of restriction for users of AAR and ARS.
Facility test calls
Allows telephone users to make test calls to access specific trunks, dual tone multifrequency
receivers, time slots, and system tones. The user dials an access code and makes the test call
to make sure the facility is operating properly. Security measures are included to prevent
unauthorized use.
Issue 6 May 2009
223
System management
Firmware download
The firmware download feature allows you to download an image from a remote or local source
into the system running Communication Manager, and use that image to reprogram the
application code of a port circuit pack. This feature makes updating firmware more cost
effective. This feature also reduces the expense of servicing the system port circuit packs
because it eliminates the need for a technician to be involved when a board is updated.
Firmware download is achieved using the TN799C CLAN interface.
Note:
Note:
Circuit packs that can be updated with the firmware download feature have a “P”
at the end of their TN number.
Five EPN maximum in MCC1 Media Gateways
Note:
Note:
This feature is for MCC1 Media Gateways when used with an S87XX Server or
DEFINITY® Server R configurations only.
This optional software feature allows customers that require high calling traffic capacities to
have from two to five expansion port networks (EPN) in a single MCC1 Media Gateway. Only
two port networks (PN) are generally available unless a specialized cable was purchased from
Avaya and work-arounds were performed in software administration to make additional carriers
function as EPNs.
When this feature is activated, Communication Manager enables administration of up to five
carriers as EPNs and no custom cables are necessary. This means that the full bandwidth of the
TDM bus is available to each carrier while still enabling the customer to have the footprint of an
MCC1 Media Gateway. This is especially appealing to call centers without IPSI/PNC
duplication, where systems can be quite large and heavily utilized.
The hardware limitation of the MCC1 Media Gateway is five port carriers. All five can be
expansion port carriers, although traffic considerations may dictate some number less than that
which is optimum. For example, a customer may choose to have three EPN carriers and two
standard port carriers.
There is only one maintenance board, which is placed in carrier A. This is the only maintenance
board in the cabinet.
224 Avaya AuraTM Communication Manager Overview
Information and reports
Note:
Only two PNs are physically supported in S87XX Server IPSI-enabled systems
when high/critical reliability options are desired. Only two PNs are physically
supported in DEFINITY Server R systems when critical/ATM Network Duplication
reliability is desired.
Note:
For more information on this feature, see your local Avaya representative.
Information and reports
l
Attendant position report
The attendant position report lists the following:
- Attendant usage
- Number of calls answered
- Total time the attendant was available to answer a new call
- Average holding time on calls answered
l
Avaya Software Compatibility Audit (ASCA) tool
The Avaya Software Compatibility Audit (ASCA) tool allows a customer to:
- Run a report on the software and firmware versions that are active on their system
- Compare their software and firmware versions with the latest Avaya releases, and
indicating the suggested upgrades
l
Blockage study report
l
Call coverage reports
The call coverage report displays measurements of the distribution of traffic offered to
call-coverage groups. Separate reports for all calls and external calls are supplied.
l
Coverage points report
The coverage points report differs based on whether all calls or external calls is selected.
For each coverage point in the group, the number of calls offered, abandoned while at that
coverage point, and overflowing to the next coverage point are listed.
l
Display ARP reports
l
Emergency and journal reports
The emergency and journal report is based on information from all crisis alerts.
l
Hunt group measurements report
l
IP reports
Issue 6 May 2009
225
System management
l
Packet error history report
Provides a 24-hour history of important packet level statistics that indirectly indicate some
LAN performance characteristics. The 24-hour history gives the ability to look back at these
measures if the trouble cleared.
l
Port network and link usage report
l
Processor occupancy report
The processor occupancy report provides summary information on how heavily the
processor is loaded.
l
Recent change history report
Allows the system manager to view or print a history report of the most recent
administration and maintenance changes on the switch. This report may be used for
diagnostic or information purposes.
l
Refresh route reports
l
Summary report
The summary report provides a performance summary of your system running
Communication Manager.
l
Tandem traffic report
The tandem traffic report provides information on facilities that serve tandem traffic.
l
Tracelog
The Tracelog, among other things, lists:
- all IP endpoint registrations
- all IP endpoint unregistrations
- all Ethernet interfaces coming into service
- all Ethernet interfaces coming out of service
These events are tagged as a new log type.
l
Traffic reports
Traffic reports show measurements in the format of switch-based reports for local or remote
access, and can be collected for subsequent analysis and reporting by adjuncts and
operation support systems using the operation support system interface protocol.
l
IP traffic measurements reports
These reports show DSP activity on IP port networks and IP media processors for specific
regions and time periods. DSP measurements are also available for H.248 gateways.
l
Voice/Network Statistics reports
The voice/network statistics reports show hourly and summary level measurement data on
packet loss, jitter, round trip delay, and data calls.
226 Avaya AuraTM Communication Manager Overview
Information and reports
l
Trunk group detailed measurements
Enhanced logging of user actions
Enhanced logging in Communication Manager servers increases logging of user actions. The
following major enhancements are included:
l
l
l
l
Modifying the Communication Manager software to log administrator activity in the System
Administration Terminal (SAT) to the Linux syslog.
Providing the ability for the system administrator to define what SAT activity is logged and
at what level, including the ability to log changed values both prior to and following a
change.
Enhancing logging of web pages.
Ensuring that all non-debugging logs use syslog and that logs can be sent to a network
syslog server of the customer’s choosing.
Parsing capabilities for the history report
The history report provides details about every data command. You can use parsing options to
limit the data returned in this report. The following table identifies the parsing options that are
available.
Note:
Note:
You can display these options by entering the command list history, then
clicking HELP or pressing F5.
Option
Description
date
Specify the month (MM) or day (MM/DD) for which to
display history data.
time
Specify the hour (HH) or minute (HH:MM) for which to
display history data.
login
Specify the login for which you wish to display history
data.
action
Specify the command action (the first word of the
command string) for which you wish to display history
data. You can view the list of available command
actions by clicking HELP or pressing F5 at the
command line.
1 of 2
Issue 6 May 2009
227
System management
Option
Description
object
Specify the command object for which you wish to
display history data.
qualifier
Specify the command qualifier for which you wish to
display history data.
2 of 2
To limit the data displayed in the history report, enter the command list history followed by
a space and the appropriate parser and, if applicable, format. Only the data for the specified
parsers will appear in the report.
You can include multiple parsers, but only a single instance of any parser (for example, you may
parse for date, time, and login, but not for date, time, and two different logins).
IP asynchronous links
See IP asynchronous links on page 176.
Malicious call trace
The New Malicious Call Identification (MCID) feature enables your server to signal to an ETSI
ISDN network when the network should record the source of an incoming call. For more
information, See Malicious call trace on page 202.
Malicious call trace logging
See Malicious call trace logging on page 202.
Music-on-hold
Automatically provides music, silence, or tone to a caller. Music lets the caller know that the
connection is still valid.
228 Avaya AuraTM Communication Manager Overview
Restriction - controlled
Local music-on-hold
The music on hold feature is supported on the G700 Media Gateway with Communication
Manager. The music source is connected to a port on the MM711 Analog Media Module. Local
music-on-hold is part of the call center functionality on the S8300 Server.
Local music-on-hold allows one music source. To use multiple music sources on a G700 Media
Gateway, you must use multiple ports on the MM771 Analog Media Module, one for each music
source.
For more information, see Installation for Adjuncts and Peripherals for Avaya Aura™
Communication Manager. Also see Administering Avaya Aura™ Communication Manager,
03-300509.
Multiple music sources
On an MCC1, SCC1, CMC1, or G600 Media Gateway, this feature allows the customer to
provide multiple distinct music sources for use with the call vectoring features, calls placed on
hole, calls awaiting pickup, and so on. By purchasing the multiple music-on-hold (also called
tenant partitioning) feature, you can have up to 100 music sources.
Many different music options can be administered to accommodate different tenants. See
Tenant partitioning on page 230.
Restriction - controlled
Allows an attendant or telephone user, with console permission, to activate and deactivate for
an individual telephone or a group of telephones, the following restrictions:
l
outward
l
total
l
station-to-station
l
termination restrictions
Issue 6 May 2009
229
System management
Scheduling
Functional scheduling in Communication Manager allows you to specify the time a command
will be executed or to specify that it should be executed on a periodic basis. Only commands
that do not require user interaction after being entered on the command line (such as list,
display, test) can be scheduled.
Security Violation Notification
See Security Violation Notification on page 206.
Station security codes
See Station security codes on page 206.
Tenant partitioning
Allows partitioning of the system in order to lease the system services and features to multiple
tenants. This provides attractive services and revenue for “virtual” landlords. It provides the
robust features of a large system at affordable rates to small business tenants. Depending on
the platform, or server, you are using, Communication Manager supports multiple partitions and
attendant groups.
Multiple attendant groups can be assigned to each partition. Stations, hunt groups, and other
endpoints assigned to a Class of Service (COS) can be partitioned. Network routing pattern
preferences also support the assigned tenant partitioning. Tenant partitioning also allows you to
assign a unique music source for each tenant partition for customers who are put on hold.
See Music-on-hold on page 228.
230 Avaya AuraTM Communication Manager Overview
Terminal Translation Initialization
Terminal Translation Initialization
See Terminal Translation Initialization on page 119.
Time of day clock synchronization through a LAN source
Customers need accurate and common time of day time source across multiple switches in a
network. This is especially important when customers are using a central Avaya Call
Management System (CMS) to report events coming from multiple servers running
Communication Manager.
The time of day clock synchronization through a LAN source feature is implemented on two
different platforms:
l
Linux
l
UNIX
Linux platforms
Communication Manager that is running on Linux-based Avaya 8XXX Servers synchronizes
time directly from a LAN source.
UNIX platforms
Communication Manager running on DEFINITY servers which use an Oryx/Pecos operating
system (proprietary UNIX-based OS) receives a command from Avaya site administration to
adjust the time. Avaya site administration is synchronized to the LAN PC's clock.
Trunk group circuits
Trunks provide the communications links between Communication Manager and other
switches, including central office switches and other premises switches. Trunks that perform the
same function are grouped together and administered as trunk groups. Trunks interface with
Communication Manager through port circuit packs.
Issue 6 May 2009
231
System management
Variable length ping
See Variable length ping on page 163.
Variable Length Subnet Mask
See Variable Length Subnet Mask on page 164.
232 Avaya AuraTM Communication Manager Overview
Chapter 20: Telecommuting and remote office
Coverage of calls redirected off-net
Coverage of calls redirected off-net (CCRON) allows calls that have been redirected to
locations outside of the switch to return to the switch for further processing.
For example, an employee that telecommutes can have two coverage paths. One coverage
path is used when the employee is in the office and the other coverage path is used when the
employee is working from home. The coverage path used from home takes a call to the
employee work telephone, and covers the call to the employee home telephone. If the
employee does not answer the call or is busy on another call, the call is redirected back to the
switch for further processing, such as coverage to voice mail.
Remote call coverage and call forwarding off-net allow calls to be redirected to a remote
location. This allows you to have calls placed to your on-site office redirected to your home
office. You can administer the system to either monitor calls and bring them back for additional
processing if not answered or to leave calls at the remote (off-net) location.
Extended user administration of redirected calls
(telecommuting access)
Extended user administration of redirected calls (also called telecommuting access) allows you
to change the lead call coverage path or forwarding extension from any on-site or off-site
location. Thus you can change the path or extension from your home office, for example.
IP Softphone and IP Agent - RoadWarrior mode
See IP Softphone and IP Agent - RoadWarrior mode on page 84.
Issue 6 May 2009
233
Telecommuting and remote office
IP Softphone and IP Agent - Shared Control mode
See IP Softphone and IP Agent - Shared Control mode on page 84.
IP Softphone and IP Agent - Telecommuter mode
See IP Softphone and IP Agent - Telecommuter mode on page 84.
IP Softphone
See Avaya IP Softphone on page 83.
Off-premises station
A trunk-data module connects off-premises private-line trunk facilities and Communication
Manager. The trunk-data module converts between the RS-232C and the DCP, and can
connect to DDD modems as the DCP member of a modem pool.
See Call redirection on page 240 and Call vectoring on page 59.
Remote access
Permits authorized callers from remote locations to access the system via the public network
and then use its features and services.There are a variety of ways of accessing the feature.
After gaining access, you hear a system dial tone, and, for system security, may be required to
dial a barrier code.
234 Avaya AuraTM Communication Manager Overview
Chapter 21: Telephony
Abbreviated Dialing
Abbreviated Dialing (AD) provides lists of stored numbers you can use to:
l
Place local, long-distance, and international calls
l
Activate features
l
Access remote computer equipment
You simply dial the list number and the one-digit, two-digit, or three-digit number associated with
the telephone number you want. The number is then automatically dialed by the system. A
frequently called number can be stored on an abbreviated dialing button that you need only
press once to make the call.
Abbreviated dialing labeling
An administrator can type personalized labels for the Abbreviated Dialing (AD) System List
entries. Users of the 2420 DCP telephone, as well as the 4600-series, 6400-series, and
8400-series display telephone sets, can administer labels for the AD softkey buttons. These
personalized labels appear on the menu display.
These personalized labels can be administered in the standard supported languages (English,
French, Italian, Spanish, and a user-defined language). If a personalized label has not been
administered for the AD system list entry, the feature button label that is downloaded to the
telephone is ADnn, where nn is the abbreviated dialing number.
Note:
Note:
This enhancement applies only to the AD system list.
Abbreviated dialing on-hook programming
On-hook programming allows users of the 2420 DCP telephone, as well as the 4600-series,
6400-series, and 8400-series telephone sets with enabled speakers, to access the
programming mode without going off-hook during available call appearances. Signaling
changes from DTMF to the S-channel, allowing the use of a longer (60 seconds) time-out
period. Signaling will remain DTMF and the current time-out period of 10 seconds will still apply
to non-display telephone sets.
Issue 6 May 2009
235
Telephony
ABCD tone support
The support of signaling messages for generating DTMF tones for the characters A, B, C, and D
is added to the H.248 gateway family. These tones are generated on both line-side and
trunk-side interfaces on the H.248 gateways for purpose of supporting signaling for our voice
adjunct applications. These applications currently include:
l
Voice Response Units (VRU) for Call Center
l
Call-me, Find-me, auto-attendant transfer, blind-transfer features for Modular Messaging
l
MLPP (Multi-Level Precedence & Preemption) feature for military/government applications
Active dialing
6400-series and 4600-series telephone sets have a dialing option where the set will send
S-channel button codes when the user presses a number on the dial pad when on-hook.
Administrable timeout on call timer
Enhances the call timer feature on the 6400-series telephones. The call timer feature measures
the duration of a call, starting a timer when the call is answered and stopping the timer when the
call is dropped.
Previously, the call timer feature displayed the duration of the call for only five seconds after the
call was dropped. The administrable timeout on call timer feature allows the user to specify how
long to display the duration of the call.
Alphanumeric dialing
See Alphanumeric dialing on page 89.
236 Avaya AuraTM Communication Manager Overview
Automatic Call Back
Automatic Call Back
Automatic Call Back (ACB) allows users who placed a call to a busy or unanswered telephone
on an internal or public network to be called back automatically when the called telephone
becomes available.
When a user activates automatic callback, the system monitors the called telephone. When the
called telephone becomes available to receive a call, the system originates the automatic
callback call. The originating party receives priority ringing. The calling party then lifts the
handset and the called party receives the same ringing provided on the original call.
Automatic Call Back allows queuing of called parties. This feature works on analog, DCP, IP
(H.323), and SIP telephones.
Automatic Call Back for analog telephones
When a person, using an analog telephone, places a call and the line is busy, an announcement
prompts the caller to enter the digit 1 to activate ACB, or to enter the digit 2 to route the call to a
hunt group extension.
Automatic hold
Allows attendants and multi-function telephone users to alternate easily between two or more
calls. For example, with automatic hold, selection of a second call automatically puts the active
call (if any) on hold and makes the second call active. This feature can be activated on a
system-wide basis only. When automatic hold is not activated, the selection of the second call
drops the first call.
Avaya video telephony solution
The Avaya Video Telephony Solution enables Communication Manager to merge a set of
enterprise features with Polycom’s video conferencing adjuncts. It unifies Voice over IP with
video, Web applications, Avaya’s video enabled IP softphone, third-party gatekeepers, and
other H.323 endpoints.
The following components are part of the Avaya Video Telephony Solution feature:
Issue 6 May 2009
237
Telephony
l
l
l
l
l
Polycom VSX3000, VSX7000 and VSX8000 conferencing systems with Release 8.03 or
later
Polycom V500 video calling systems
Polycom RMX 2000 video bridge is supported. For more specific information on the RMX
2000 product, see: http://www.polycom.com/usa/en/products/products.html
Polycom MGC video conferencing bridge platforms with Release 8.0.1 are supported, but
with some limitations. Release 7.5 of the MGC is not supported.
Third-party gatekeepers, including Polycom Path Navigator
You also need a system running Communication Manager Release 3.1.3 or later, and Avaya IP
Softphone Release 5.2 with video integrator.
For more information, see Avaya Aura™ Communication Manager Feature Description and
Implementation, 555-245-205.
Bellcore calling name ID
Allows the system to accept calling name information from a Local Exchange Carrier (LEC)
network that supports the Bellcore calling name specification. The system can send calling
name information in the format if Bellcore calling name ID is administered. The following caller
ID protocols are supported.
l
Bellcore (default) - US protocol (Bellcore transmission protocol with 212 modem protocol)
l
V23-Bell - Bahrain protocol (Bellcore transmission protocol with V.23 modem protocol).
Bridged call appearance - multi-appearance telephone
Allows calls made to or from a primary telephone extension to be handled from more than one
telephone. A bridged call appearance is set up by administering a primary extension and the
button number associated with it on a multi-lamp button on another telephone. This feature is
most often used by secretaries or assistants who answer or handle calls to the primary
extension (an executive, for example).
When the primary extension receives a call, the bridged call appearance flashes or rings on all
telephones administered with this feature. The call can be answered by anyone having a
telephone with this feature and handled as if the primary extension user was answering it. The
maximum number of bridged appearances is 64.
238 Avaya AuraTM Communication Manager Overview
Bridged call appearance - single-line telephone
Bridged call appearance - single-line telephone
Allows single-line telephones users to have a bridged appearance on a multi-appearance
telephone.
Call coverage
Call coverage provides automatic redirection of calls that meet specified criteria to alternate
answering positions in a call coverage path. A coverage path can include any of the following:
l
a telephone
l
an attendant group
l
a Uniform Call Distribution (UCD) hunt group
l
a Direct Department Calling (DDC) hunt group
l
an Automatic Call Distribution (ACD) hunt group
l
a voice messaging system
l
a Coverage Answer Group (CAG) established to answer redirected calls
Alphanumeric field designation
In addition to numeric designations for key system lists and groups of related information, the
system administrator can specify alphanumeric designations, 0-15 characters in length, for the
following:
l
abbreviated dial lists
l
abbreviated dial groups
l
call pickup groups
l
call routing patterns
Changeable coverage paths
Changeable coverage paths allows the end user to modify the coverage points by using a
feature access code (FAC).
Issue 6 May 2009
239
Telephony
Directory
The directory feature allows users with display-equipped telephones to access the system
database, use the touch-tone buttons to enter a name, and retrieve an extension number from
the integrated directory. The directory contains the names and extensions assigned to all
telephones on the system. The directory feature can also handle Russian names.
The integrated directory enables speed dial for both internal and external numbers while
decreasing the number of dedicated buttons for speed dial. It also provides for a common
directory across all endpoints. The integrated directory is linked to an external Lightweight
Directory Access Protocol (LDAP) directory.
Enhanced coverage and ringback for logged off IP/PSA/TTI
stations
A field on the system-parameter customer-options screen, called Don’t Answer Criteria For
Logged Off IP/PSA/TTI Stations?, allows a customer to choose how to handle non-registered
stations. Possible entries for this field are y or n.
l
If this field is set to n, call forwarding follows the busy criteria.
l
If this field is set to y, call forwarding follows the don’t answer criteria.
Time of day
This feature allows a user to have multiple coverage paths depending on the time of day, and
day of the week.
Call redirection
Call forward busy/do not answer
Allows calls to be forwarded when the called extension is busy or when the call is not answered
after an administrable interval. If the extension is busy, the call forwards immediately. If the
extension is not busy, the incoming call rings the called extension, then forwards only if it
remains unanswered longer than the administered interval.
240 Avaya AuraTM Communication Manager Overview
Call redirection
Call forwarding all calls
Allows calls to be forwarded to an internal extension, external (off-net) number, an attendant, or
an attendant group. You can include an access code or special characters, like pause
characters, in the forwarding destination.
Call forwarding enhancements
Enhancements to the Call Forward feature give users the following capabilities:
l
Designate different preferred destinations for calls that originate from internal and external
callers
l
Separate call forwarding/busy don’t answer calls
l
Three different ways to forward incoming calls
- Enhanced call forwarding unconditional (ECFU)
- Enhanced call forward busy (ECFB)
- Enhanced Feature no reply (ECFNR)
Chained call forwarding
This feature allows calls to be forwarded up to 10 hops (each calling station is considered to be
one hop), the default value being 3 hops. Chained call forwarding allows mixed use of standard
Call forwarding and Enhanced call forwarding. You can use the coverage path of the last
forward-to station in the chain for coverage after forwarding.
Enhanced Redirection Notification
The activation of redirection features at a user's station is indicated through visual display or
through a special dial tone (if the station is not equipped with a display). This feature works on
DCP and IP (H.323) telephones, but not on SIP telephones and attendant consoles.
Call forwarding override
Allows the user at the forwarded-to extension to override call forwarding and either initiate a call
or transfer a call back to the forwarded-from extension.
Issue 6 May 2009
241
Telephony
Send All Calls and Call Forwarding Override
The Send All Calls and Call Forwarding (SAC/CF) Override feature overrides active rerouting.
This feature overrides these active rerouting settings:
l
Send All Calls (SAC)
l
Call Forwarding (CF) all
l
Enhanced Call Forwarding (ECF) unconditional
The SAC/CF override feature depends on call initiation. On enabling SAC/CF override, the call
may:
l
Execute override - ring called station
l
Cancel the override
l
Display a message and wait for further input
Call redirection intervals
Avaya AuraTM Communication Manager allows the system administrator to specify the number
of times that a call rings at each call coverage point before the call proceeds to the next
coverage point.
Call Log enhancements
The Log Forwarded Call operation creates missed call log entries for calls that are redirected by
Call Forward All, Send All Calls, or Goto Cover. You have the option to decide whether to log
calls forwarded to another party on the forwarding telephone as missed calls or not. Call
Forward, Enhanced Ca all Forward, Coverage and the interactions around these [Send All Calls
(SAC) button, Goto Cover button, Do Not Disturb (DND)] are covered.
Call park
Allows you to put a call on hold and then retrieve a call from any other telephone on the system.
This is helpful when you are on a call and need to go to another location for information. It also
allows you to answer a call from any telephone after being paged by a telephone user or an
attendant.
242 Avaya AuraTM Communication Manager Overview
Call pickup
Call pickup
Along with directed call pickup, allows you to answer calls for other telephones within your
specified call pickup group. Directed call pickup allows you to pick up any call on the system.
With this feature, you do not have to leave your telephone to answer a call for a nearby
telephone. You simply dial an access code or press a call pickup button.
Enhanced Call Pickup Alerting
Call Pickup Alerting provides display of calling and called party information for all members of
the pickup group and administrable alerting options. This feature works on DCP and IP (H.323)
telephones but not on SIP telephones.
Group call pickup
Allows you to dial a feature access code (FAC) and a pickup group number to answer a call
from a different group. For example, marketing would be able to pickup calls in the sales group
when the sales group is unavailable. This feature is ideal for offices that are not divided by
partitions and generally have the departments on the same floor.
Caller ID on analog trunks
See Caller ID on analog trunks on page 143.
Caller ID on digital trunks
See Caller ID on digital trunks on page 143.
Circular station hunt group
See Circular station hunt group on page 62.
Issue 6 May 2009
243
Telephony
Conferencing
See Conferencing on page 73.
Consult
Allows a covering user, after answering a call received through call coverage, to call the called
party for private consultation. Consult can be used to let a covering user ask the principal if they
want to speak with the calling party.
Coverage callback
Allows a covering user to leave a message for the called party to call back the person who
called.
Coverage incoming call identification
Allows multi-appearance telephones users without a display in a coverage answer group to
identify an incoming call to that group.
Disconnecting unanswered calls
Disconnects unanswered outgoing calls after a predetermined amount of time. When any of the
following timers expire during an outgoing local, toll, or international call attempt, the switch
disconnects the call and applies busy tone, which may or may not be followed by howler tone:
l
Pre-dialing and interdigit timer
l
Outgoing seizure acknowledge timer
l
Answer supervision timer
l
60-, 90-, and 120-second no-answer disconnect timers, based on ARS call type
244 Avaya AuraTM Communication Manager Overview
Distinctive ringing
l
120-second timer used for calls without a call type, such as calls to trunk access codes
Distinctive ringing
Rings or activates alerting on your telephone in such a way that you are aware of the type of
incoming call before answering it. This feature operates in a Distributed Communications
System (DCS) environment the same as it does within a single system. In Communication
Manager, you can also administer distinctive ringing on a per-tenant basis if you use partitioning
in your system.
By default, internal calls are identified by a 1-burst ringing pattern, external calls by a 2-burst
ringing pattern, and priority calls by a 3-burst ringing pattern. You can administer these patterns.
Maintain external ring tone after internal transfer
When a call from outside the Avaya AuraTM Communication Manager system is transferred
internally from one Avaya AuraTM Communication Manager station to another, ringing that
indicates that the original call is from an external source is maintained. In other words, the
transferred-to telephone rings to indicate an external call.
Edit dialing
Edit Dialing feature allows an end-user to pre-dial a number when the telephone is on-hook.
During the pre-dial phase the user can edit the digits of a dialed number. The number is dialed
when the user goes off-hook (lifts handset or presses the speaker button) or presses the send
soft key. Edit Dialing is supported by Spice telephones, which are telephones from the 96xx
Series with the protocol version H.323 (9620 Lite, 9620 Color, 9650 Color). To use this feature,
the software version must be Spice 3.0 or later. SIP telephones are not supported by this
feature.
Emergency calls from unnamed IP endpoints
With the Emergency calls from unnamed IP endpoints feature, an IP telephone can register
without an extension number. The Emergency calls from unnamed IP endpoints feature places
the IP telephone into Terminal Translation Initialization (TTI) service. Users can dial a feature
Issue 6 May 2009
245
Telephony
access code (FAC) to either associate an extension number with a telephone, or to dissociate
an extension number from a telephone.
If Avaya AuraTM Communication Manager is appropriately administered, a user can use an IP
telephone that is in TTI service to make emergency or other calls.
The Emergency Calls from Unnamed IP Endpoints feature requires IP telephone software R2.3
or later. IP telephone software R2.3 or later requires TN799C hardware vintage 3 or later circuit
packs. TN799C hardware vintage 1 and 2 circuit packs do not work with IP telephone software
R2.3. All versions of TN799DP circuit packs are compatible with IP telephone software R2.3.
Therefore, the Emergency Calls from Unnamed IP Endpoints feature requires TN799DP or
TN799C hardware vintage 3 or later circuit packs.
Enhanced abbreviated dialing
Supplements abbreviated dialing by providing one enhanced number per system. Enhanced
number lists can contain any number or dial access code. System administrators designate
privileges for group number lists, system number lists and enhanced number lists. With
privileged lists, users can access otherwise restricted numbers (for example, telephones
without long-distance access can be programmed to access specified long-distance numbers).
The Avaya 8XXX Server supports 20,000 entries within the enhanced abbreviated dialing
system list. This second enhanced abbreviated dialing list doubles the capacity to from 10,000
entries to 20,000 entries.
Future increases to the enhanced abbreviated dialing list can be performed easily by increasing
the number of lists. Increasing the number of lists increases the overall capacity by multiples of
10,000 entries.
Enhanced telephone display
The enhanced telephone display feature allows you to choose the character set that you want to
see in Avaya AuraTM Communication Manager softkeys and display telephones. In addition to
the standard Roman character set, you can choose either the Katakana or characters used for
most European languages.
246 Avaya AuraTM Communication Manager Overview
Enterprise Mobility User
Enterprise Mobility User
Use the Enterprise Mobility User (EMU) feature to associate the features and permissions of the
primary telephone of a user to a telephone of the same type anywhere within the customer
enterprise. With EMU, the Class of Restriction (COR) and Class of Service (COS) permissions
of the primary telephone are not associated with other telephones in the system.
Note:
Any telephone that is not the primary telephone is referred to as the visited
telephone. Any server that is not the home server of the primary telephone is
referred to as the visited server.
Note:
The following list outlines the requirements for the EMU feature:
l
l
l
l
QSIG must be the private networking protocol in the network of Communication Manager
systems. This requirement also includes QSIG MWI.
Communication Manager software must be running on the home server and all visited
servers.
All servers must be on a Linux platform. EMU is not supported on DEFINITY servers.
The visited telephone must be the same model type as the primary telephone to enable a
optimal transfer of the image of the primary telephone. If the visited telephone is not the
same model type, only the call appearance (call-appr) buttons and the message waiting
light are transferred.
l
All endpoints must be self-designating terminals.
l
Uniform Dial Plan (UDP).
Calls made from a visited telephone can be processed by either the home server or the visited
server. Which server processes the call depends on how the user originates the call. The home
server processes any calls that are a result of a user depressing one of the buttons that were
downloaded to the visited telephone. The visited server processes any calls that are placed on
the visited telephone using the dial pad.
Enterprise Mobility User enhancements
Enhancements are made to the Enterprise Mobility User (EMU) feature for release 4.0:
l
l
Enable EMU while mobile within a single Avaya AuraTM Communication Manager server
domain.
Permit visited station’s principal user to use Extension to Cellular OPTIM application
during a registered EMU period at that station.
You can administer the amount of time that must elapse before a visitor can log in as EMU after
making emergency calls from an EMU-activated station.
Issue 6 May 2009
247
Telephony
Enterprise Wide Licensing
Enterprise Wide Licensing (EWL) is a technology within Avaya AuraTM Communication
Manager release 3.0 and a Avaya AuraTM Communication Manager license file. EWL is used to
partially support a developing offer known as Enhanced Software License Program (ESLP).
ESLP gives customers the ability to bulk purchase and then share license capacities across
multiple locations.
Note:
Note:
Launch of the ESLP offer will be announced at a later date.
Go to cover
Allows users who call another internal extension to send the call directly to coverage.
Hold
Allows you to disconnect from a call temporarily, use your telephone for other call purposes, and
then return to the original call.
Intercom - automatic answer
Automatic answer intercom calls (auto answer ICOM) allows a user to answer an intercom call
within the intercom group without pressing the intercom button. Auto answer ICOM works with
digital, BRI, and hybrid telephones with built-in speaker, headphones, or adjunct speakerphone.
Internal automatic answer
Allows specific telephones to answer incoming internal calls automatically. This feature is
intended for use with telephones that have speakerphones or headsets. You simply press an
internal automatic answer feature button, and calls are automatically answered when the
248 Avaya AuraTM Communication Manager Overview
Last number dialed
telephone is idle. Internal and Distributed Communications System (DCS) calls can be
answered using automatic answer, but only attendants can use automatic answer to answer
external calls directed to the attendant.
Last number dialed
Allows you to automatically redial the last number dialed. The system saves the first 24-digits of
the last number dialed, whether the call attempt was manually dialed or dialed using
abbreviated dialing. When you press the last number dialed button or dial the last number dialed
feature access code, the system places the call again.
Local call timer automatic start/stop
Automatically starts the local timer of a 6400-series telephone when a call is received. The timer
is stopped automatically when a call is ended. When a call is placed on hold the timer continues
to run, but is not displayed. When the call comes off hold, the total elapsed call-time displays.
Long hold recall
Visual and audible warnings are sent to the telephone where a call has been on hold past a
specified period of time. Both visual and audible warnings are used if the telephone is on-hook.
If the telephone is off-hook, a “priority ring” is used. This is an optional feature at the system
level.
Manual originating line service
Connects single-line telephone users to the attendant automatically when the user lifts the
handset. The attendant number is stored in an abbreviated dialing list. When the telephone user
lifts the handset, the system automatically routes the call to the attendant using the hot line
service feature.
Issue 6 May 2009
249
Telephony
Misoperation handling
Defines how calls are handled when a misoperation occurs. A misoperation is when calls are
left on hold when the controlling station goes on hook.
For example, a misoperation can occur under either of the following conditions:
l
l
If you hang up prior to completing a feature operation (in some cases, hanging up
completes the operation, as in call transfer). If, for example, you place a call on hold, begin
to transfer the call, dial an invalid extension number, and then hang up, that is a
misoperation.
When the system enters night service while attendant consoles have calls on hold.
The system administrator can alter the standard misoperation handling to ensure that an
external caller is not left on hold indefinitely, or dropped by the system after a misoperation with
no way to reach someone for help.
Note:
This feature is required only in France and Italy, but it can be used at any location
where the feature has been turned on.
Note:
Multiappearance preselection and preference
Provides options for placing or answering calls on selected call appearances.
l
l
l
Ringing appearance preference automatically connects you to the incoming ringing call
when the user picks up the handset.
Idle appearance preference automatically connects you to an idle appearance.
Preselection allows the user to manually select an appearance. Preselection is used, for
example, when you want to reconnect with a held call or activate a feature.
Preselection can be used with a feature button. For example, if you press an abbreviated
dialing button, the call appearance is automatically selected and, if you pick up the handset
within five seconds, the call is automatically placed. The preselection option overrides both
of the other preference options.
250 Avaya AuraTM Communication Manager Overview
Multiple level precedence and preemption
Multiple level precedence and preemption
Multiple level precedence and preemption (MLPP) is an optional group of features that provide
users the ability to interface to and operate in a Defense Switched Network (DSN). The DSN is
a highly secure and standards-based communication system of the US Government
Department of Defense (DoD).
! CAUTION:
MLPP is currently designed to meet only DoD GSCR requirements for connection
to a DSN by federal, state, or local government agencies. As such, MLPP is not
currently designed for use in commercial enterprise environments. Activation of
this feature in any other kind of network environment could result in unexpected
and unwanted feature operations.
CAUTION:
The MLPP features allow users to request priority processing of their calls during critical
situations. The MLPP features include:
l
Announcements for precedence calling
l
Dual homing
l
End office access line hunting
l
Line load control
l
Precedence call waiting
l
Precedence calling
l
Precedence routing
l
Preemption
l
Worldwide numbering and dialing plan
Announcements for precedence calling
In certain situations, precedence calls are blocked because of unavailable resources or
improper use. When this occurs, recorded announcements are used to identify what went
wrong. The announcements used for MLPP include:
l
Blocked precedence call
l
Unauthorized precedence level attempted
l
Service interruption prevented call completion
l
Busy, not equipped for preemption or precedence call waiting
l
Vacant code
Issue 6 May 2009
251
Telephony
Dual homing
Dual homing allows a user to dial a telephone number and, if the initial route is unavailable,
have the call route to its destination over alternate facilities.
End office access line hunting
End office access line hunting automatically hunts for an idle trunk over end office access lines,
based on the precedence level of the call.
Line load control
Line load control is a feature that restricts a predefined set of station users from originating calls
during a crisis or emergency. Through administration, users are assigned to a line load control
level based on their relative importance. When an emergency occurs, the administrator
manually enables the feature to restrict calling by users of lower importance. When the
emergency is over, the administrator manually disables the feature.
For example, if a situation occurs that threatens national defense, station users in the defense
department will not be restricted from originating calls, but stations in other departments, such
as the accounting department, will be restricted. When the crisis is over, the system can be
returned to normal operation by the administrator.
Precedence call waiting
After a precedence call is routed, the called party may already be busy on another call.
Precedence call waiting allows the caller to “camp on” to the line of the called party and wait for
the user to answer the call. The caller hears a special ringback tone and the called party hears
a call waiting tone.
Depending on the type of telephone being used, the called party can put the current call on hold
and answer the call, or the called party must hang up on their current call to answer the
incoming call.
Precedence calling
Precedence calling is the centerpiece of the MLPP features. Precedence calling allows users,
on a call-by-call basis, to select a level of priority for each call based on their need and
252 Avaya AuraTM Communication Manager Overview
Multiple level precedence and preemption
importance (rank). The call receives higher-priority routing, whether the call is local or going
around the world.
Users may access five levels of precedence when placing calls:
l
Flash Override (the highest precedence level)
l
Flash
l
Immediate
l
Priority
l
Routine (the default, and lowest precedence level)
Each station user is administered with a maximum precedence level. The more important or
higher in rank of the user, the higher the precedence level. Users cannot originate calls at
precedence levels higher than their maximum administered level. Non-MLPP calls are treated
as routine level precedence calls.
Precedence routing
When precedence calls are destined for other switches in a private network, the precedence
routing feature is used to route the calls. The precedence routing feature routes calls based on
three main criteria:
l
Routing based on the destination number
l
Routing based on the precedence level
l
Routing based on the time of day
These routing criteria are administrable and can be changed as required. Two related features
are dual homing and end office access line hunting.
Preemption
Preemption works with the precedence routing feature to further extend the call routing
capabilities of the MLPP features. Preemption, when allowed through administration, can
actually tear down an existing lower priority call in order to complete a more important
precedence call. Even non-MLPP calls are treated as routine level precedence calls and can be
preempted.
When this occurs, the callers on the existing call hear a tone indicating that the call is about to
be preempted. The callers have three seconds to end the call before the call is automatically
disconnected. After the existing call is disconnected, the new call is placed using preempted
facility.
Issue 6 May 2009
253
Telephony
Worldwide numbering and dialing plan
The worldwide numbering and dialing plan (WNDP) feature allows Avaya AuraTM
Communication Manager to conform to the standard numbering system established by the
Defense Communications Agency (DCA). WNDP defines its own format for the precedence
dialing. The capability to operate with this numbering plan must be incorporated into all new
switches introduced into the Defense Switched Network (DSN).
Night service
There are five night service features:
l
l
l
l
l
Hunt group night service allows an attendant or a split supervisor to assign a hunt group or
split to night service mode. All calls for the hunt group then are redirected to the hunt group
designated night service extension. When a user activates hunt group night service, the
associated button lamp lights.
Night console service directs all calls for primary and daytime attendant consoles to a night
console. When a user activates night console service, the night service button for each
attendant lights and all attendant-seeking calls (and calls waiting) in the queue are directed
to the night console. To activate and deactivate this feature, the attendant typically presses
the night button on the principal attendant console or designated console.
Night station service directs incoming calls for the attendant to designated extensions.
Attendants can activate night station service by pressing the night button on the principle
console if there is not an active night console. If the night station is busy, calls (including
emergency attendant calls) receive a busy tone. They do not queue for the attendant.
Trunk answer from any station allows telephone users to answer all incoming calls to the
attendant when the attendant is not on duty and when other telephones have not been
designated to answer the calls. The incoming call activates a gong, bell, or chime and a
voice-terminal user dials an access code to answer the call.
Trunk group night service allows an attendant or a designated telephone user to
individually assign a trunk group or all trunk groups to the night service mode. Specific
trunk groups individually assigned to the service are in Individual trunk night service mode.
Calls coming into these trunk groups are redirected to designated night service
extensions. Incoming calls on other trunk groups are processed normally.
Enhanced night service
Avaya AuraTM Communication Manager informs a voice mail system (VMS) that it is in night
service, allowing the VMS to perform different actions and call handling for out-of-hours
254 Avaya AuraTM Communication Manager Overview
License modes
operation. For example, the VMS may be administered to provide recorded announcements
after hours. The enhancement is made to the mode code voice mail interface.
License modes
The three modes of license operation for your system are:
l
License-normal mode
l
License-error mode
l
No-license mode
License-normal mode
The license-normal mode is the desired mode of operation of a stable system. During this mode
of operation, a license is properly installed, the license contains a serial number that matches
the processors, the license is not expired, and feature usage does not exceed limits.
License-error mode
The license-error mode is a warning mode. During this mode, call processing is supported, the
system declares a major alarm, and a 6-day countdown timer is running. If this timer is allowed
to expire, the no-license mode is invoked.
The license-error mode is entered as a result of one of the following conditions:
l
l
l
l
l
l
The serial number of the active processor does not match the license file.
The standby processor cannot be contacted or the serial number of the standby processor
does not match the serial numbers in the license.
The license has expired.
Feature usage exceeds limits. For example, there are more ports translated than permitted
by the port limit in the license.
A WAN spare or survivable remote processor enters License-Error mode when it is
providing primary service.
A switch has initialized after a software upgrade and a new license has not yet been
installed.
Issue 6 May 2009
255
Telephony
The license-error mode is cleared by correcting the error that caused the system to enter into
license-error mode, or by installing a valid license that is consistent with the configuration of the
switch.
Limit the number of concurrent calls
A typical Avaya AuraTM Communication Manager digital or IP station has three call
appearances.
l
l
The first two call appearances are for receiving incoming calls.
The last call appearance is reserved for call origination or receiving priority calls if the first
two call appearances are active.
When the Limit the Number of Concurrent Calls (LNCC) configuration is administered, the
LNCC feature allows the station user to limit the number of concurrent calls to one call.
LNCC uses a feature access code or a programmed button. A visual display or audio feedback
is provided that indicates if LNCC is active.
A call to a station that is LNCC-busy is treated like a call to a normal busy station. The call
follows the call coverage path for status busy if administered or hunts to the hunt-to station if
administered.
No-license mode
The no-license mode is a state in which all new call originations, except alarm calls and calls to
an administered emergency number, are denied. All incoming calls, except calls to an
administered number, are also denied.
The no-license mode state is entered as the result of one of the following reasons:
l
No license is installed in the system.
l
The License-Error timer has expired.
l
l
A survivable remote processor detects a port board in its port network other than an
Expansion Interface board.
A reset system 3 preserve-license command is executed and the offer category in
translation does not match the offer category in the license.
Starting with Avaya AuraTM Communication Manager release 2.1, no-license mode not only
protects customers from loss of call processing, but also provides software copy protection. The
result of no-license mode is an error message on telephone displays, and blocked system
administration.
256 Avaya AuraTM Communication Manager Overview
Personalized ringing
No-license mode is cleared by correcting the error that caused the system to enter into
no-license mode, or by installing a valid license that is consistent with the configuration of the
system.
Personalized ringing
Allows users of certain telephones to uniquely identify their own calls. Each user can choose
one of a number of possible ringing patterns. The eight ringing patterns are tone sequences
consisting of different combinations of three tones. With this feature, users working closely in
the same area can each specify a different ringing pattern in order to better identify their own
calls.
Posted messages
In most situations, after a few rings when no one answers a call, the calling party usually hears
an announcement saying that the called party is not available and to please leave a message.
At this point, the calling party has no clue when the called party would return the call.
The posted messages feature provides Avaya AuraTM Communication Manager users with the
capability of indicating the reason of their unavailability to calling parties. The system provides
30 messages for a user to choose from, such as “on vacation,” or “at lunch.” Of the 30
messages, 15 messages are fixed system messages, and the remaining 15 messages are
administrable (custom messages). After a user has chosen one of the messages and thus
activated the feature, the message is immediately sent to calling parties who have terminal
displays.
The system provides two ways to activate/deactivate this feature: using button pushes and
feature access codes. The system allows users to use the feature access codes from their own
display telephone, from another station/attendant, or from a remote access trunk.
Priority calling
Allows you to ring another telephone with a distinctive signal that tells the called party the
incoming call requires immediate attention. The called party can then handle the call
accordingly. You activate priority calling by dialing a priority calling access code or pressing a
feature button, followed by the extension number. You can use priority calling only if your
telephone has been administered with the required class of service.
Issue 6 May 2009
257
Telephony
Pull transfer
Allows either the party who was originally called, or the party to whom the held call will be
transferred, to complete the transfer. This is a convenient way to connect a party with someone
better qualified to handle the call. Attendant assistance is not required and the call does not
have to be redialed. It interfaces with satellite workstations through TGU/TGE trunks and is
always available for calls that use TGU/TGE trunks.
Recall signaling
Recall signaling allows the user of an analog station to place a call on hold, use the telephone
for other call purposes, and then return to the original call.
Recorded telephone dictation access
Allows telephone users, including remote access and incoming tie trunk users, to access
dictation equipment. The dictation equipment is accessed by dialing an access code or
extension number. The start/stop function can be voice or dial controlled. Other functions such
as initial activation and playback are controlled by additional dial codes.
Reset shift call
If a called number is busy and does not have coverage, or the called number and the coverage
are both busy, you have an opportunity to replace the last digit that was entered. This allows
you to call another extension without having to hang up and redial. Reset shift call is a feature
that is active for station to station (internal) calls and for private network calls. The private
network trunks must signal busy using out-of-band signaling.
258 Avaya AuraTM Communication Manager Overview
Ringback queuing
Ringback queuing
Places calls in an ordered queue (first in, first out) when all trunks are busy. The telephone user
who is trying to make a call is automatically called back when a trunk becomes available, and
hears a distinctive three-burst signal when called back.
Ringer cutoff
Allows the user of a multi-appearance telephone to turn audible ringing signals on and off.
Visual alerting is not affected by this feature. When this feature is enabled, only priority
(three-burst) ring, redirect notification, intercom ring, and manual signaling ring at the
telephone. Internal and external calls do not ring.
Ringing - abbreviated and delayed
Allows you to manually or automatically assign one of four ring types to each call appearance
on a telephone. Whatever treatment you assign to a call appearance is automatically assigned
to each of its bridged call appearances.
Ringing options
Provides multi-appearance telephone users with different ringing patterns. This feature primarily
affects audible ringing for calls directed to telephones that are off hook, or calls directed to idle
and active CALLMASTER telephones.
Send all calls
Allows users to temporarily direct all incoming calls to coverage regardless of the assigned
call-coverage redirection criteria. Covering users can temporarily remove their telephones from
the coverage path. The feature is activated and deactivated via a button or access code.
Issue 6 May 2009
259
Telephony
Special dial tone
Provides the ability to play a special dial tone whenever an analog set is not able to receive
calls. When such conditions as call forward all calls, call forward busy/no answer, send all calls,
or do not disturb are activated on a telephone set, a special dial tone lets you know that you
cannot receive any calls.
Station hunting
Routes calls made to a busy extension to another extension. To use station hunting, you create
a station hunting chain that governs the order in which a call routes from one extension to the
next when the called extension is busy. Each extension in the chain links to only one
subsequent extension. However, an extension may be linked from any number of extensions.
Station hunt before coverage
This feature changes the interaction that occurs between station hunting and call coverage.
Station hunt before coverage causes a call going to a busy station to go through a station
hunting process before going to coverage. If all the stations in the hunt group are busy, the call
will go to the coverage path.
Station self display
Station self display shows the extension number of the telephone set when a user either dials
the feature access code while off-hook, or depresses the INSPECT button when on-hook. The
dialed number will be displayed once the user starts to dial. This feature is helpful to people who
move from one desk to another while they are working. This feature is also used by
maintenance personnel to ensure that an extension number is correctly administered.
260 Avaya AuraTM Communication Manager Overview
Station used as a virtual extension
Station used as a virtual extension
Allows a customer to assign multiple, individual, virtual extensions to one physical telephone.
The physical telephone must be analog and on the local switch. The administrator can set each
virtual extension with a unique ring pattern to identify the extension for which the incoming call
is intended. For example, an administrator could assign three virtual extensions, each with a
unique ring pattern, to a single telephone shared by three roommates in a college dormitory.
This feature affects incoming calls only; all outgoing calls are associated with the physical
extension.
Support for the Hewlett Packard DL380G2 server
Avaya AuraTM Communication Manager is supported on Hewlett Packard (HP) DL380G2
servers in an S87XX IP-PNC system configuration (an S87XX Server with a G600 Media
Gateway).
Team button
The Team Button feature monitors members of a team of stations. A team is a virtual set of
stations. Members of a team can be any station type with multiple call appearance displays and
administrable feature buttons. Analog stations, BRI stations, SIP telephones, X-ports, and
X-mobiles are allowed as valid monitored stations, but not as monitoring stations. Attendant
consoles are not allowed as monitoring stations or monitored stations. The assignment of a
Team Button on a station brings it automatically into the virtual team.
The team button, using its speed dial function, can override a rerouting caused by active Send
All Calls (SAC), Call Forwarding (CFWD) all, or Enhanced Call Forward (ECF) unconditional.
When the team button is pressed on the monitoring station and the monitored station has a
direct rerouting active (SAC, CFWD all, ECF unconditional), the call may do any of the
following, depending upon administration:
l
Be rerouted
l
Ring the monitored station
l
Display a message asking the caller whether to ring the monitored station or reroute the
call
The number of team buttons per station is 31. However, the maximum number stations that can
monitor any one station is 15. Certain enhancements are made to team button display and
Issue 6 May 2009
261
Telephony
execution functions. For more information, see Avaya Aura™ Communication Manager Feature
Description and Implementation, 555-245-205.
Telephone display
Provides multi-appearance telephone users with updated call and message information. This
information is displayed on a display-equipped telephone. The information displayed depends
upon the display mode selected by the user. Information that allows personalized call answering
is available on many calls.
Users may select any of the following as the display message language: English (default),
French, Italian, or Spanish. In addition, messages can be administered on the system in a fifth
language. The language for display messages is selected by each user.
ISO 8859-1 encoding compatibility
ISO 8859-1 encoding compatibility enables Avaya AuraTM Communication Manager to support
correct display of names in languages that use the Latin1 set of symbols (also referred to as
extended ASCII). Latin1 covers most West European languages, including:
l
French (fr)
l
Spanish (es)
l
Catalan (ca)
l
Basque (eu)
l
Portuguese (pt)
l
Italian (it)
l
Albanian (sq)
l
Rhaeto-Romanic (rm)
l
Dutch (nl)
l
German (de)
l
Danish (da)
l
Swedish (sv)
l
Norwegian (no)
l
Finnish (fi)
l
Faroese (fo)
l
Icelandic (is)
262 Avaya AuraTM Communication Manager Overview
Telephone self administration
l
Irish (ga)
l
Scottish (gd)
l
English (en)
Telephone self administration
The telephone self administration capability allows you to program feature buttons on the
telephone yourself.
Temporary bridged appearance
Allows multi-appearance telephone users in a terminating extension group or personal central
office line group to bridge onto an existing group call. If a call has been answered using the call
pickup feature, the originally called party can bridge onto the call. This feature also allows a
called party to bridge onto a call that redirects to coverage before the called party can answer it.
Terminating extension group
Allows an incoming call to ring (either audible or silent alerting) as many as four telephones at
the same time. Any user in the group can answer the call. Any telephone can be administered
as a group member. Only a multi-appearance telephone can be assigned a feature button with
an associated status lamp, however.
The feature button allows the user to select a terminating extension group call appearance for
answering or bridging onto an existing call but not for call origination. For example, a
department in a large store might have three telephones. Anyone in the department can answer
the call. The salesperson most qualified to answer the call can bridge onto the call.
Any telephone can be assigned as a TEG member; however, only a multi-appearance
telephone can be assigned a TEG button with associated status lamp. The TEG button allows
the telephone user to select a TEG call appearance for answering or for bridging onto an
existing call. The TEG members are assigned on an extension number basis. Call reception
restrictions applicable to the group are specified by the group class of restriction (COR). The
group COR takes precedence over an individual member’s COR. When a Terminating
Extension Group (TEG) receives an incoming call, TEG’s primary Class of Restrictions (COR)
should be considered for the calling/called restrictions. The members could all be termination
Issue 6 May 2009
263
Telephony
restricted but still receive calls if the group is not restricted. For the Terminating Extension
Group screen, see Avaya Aura™ Communication Manager Screen Reference, 03-602878.
Time of day routing
Provides the most economical routing of ARS and AAR calls. This routing is based on the time
of day and day of the week that each call is made. Up to eight TOD routing plans may be
administered, each scheduled to change up to six times a day for each day in the week.
This allows you to take advantage of lower calling rates during specific times of the day and
week. In addition, companies with locations in different time zones can use different locations
that have lower rates at different times of the day or week. This feature is also used to change
patterns during the times an office is closed in order to reduce or eliminate unauthorized calls.
Timed call disconnection for outgoing trunk calls
This feature provides the capability to automatically disconnect an outgoing trunk call after an
administrable amount of time. Warning tones are applied to all parties on the call prior to the
disconnection.
The amount of time that can elapse before the trunk is dropped can be specified, and can vary
between 2-999 minutes. If the timer field is blank, which is the default value, then the feature is
disabled and the trunk will not be automatically disconnected.
Timed call disconnection applies to all outgoing trunk calls initiated by a party belonging to a
specified Class of Restriction (COR).
Prior to disconnecting the trunk, warning tones are applied to all parties on the call. The first
warning tone occurs when one minute is remaining on the call. The second warning tone occurs
when 30 seconds are remaining on the call.
Transfer
Allows telephone users to transfer trunk or internal calls to other telephones within the system
without attendant assistance. This feature provides a convenient way to connect a party with
someone better qualified to handle the call.
264 Avaya AuraTM Communication Manager Overview
Transfer
Abort transfer
Allows a user to abort a transfer attempt by pressing a non-idle line appearance. The call being
transferred would be taken off a transfer-type hold and be put on a traditional hold. The transfer
will also be aborted when you hang up (going on-hook), unless transfer upon hang-up is
activated on the switch. This is an optional feature at the system level.
Transfer - outgoing trunk to outgoing trunk
Allows a user or attendant to initiate two or more outgoing trunk calls and then transfer the
trunks together. The transfer operation removes the original user from the connection and
conferences the outgoing trunks. Alternatively, the controlling party can establish a conference
call with the outgoing trunks and then drop out of the conference, leaving only the outgoing
trunks on the conference. This is an optional enhancement to trunk-to-trunk transfer and
requires careful administration and use. DCS trunk turnaround may be a safer alternative to this
feature.
Transfer recall
Returns the unanswered transfer calls back to the person who transferred the call. Transfer
recall uses a priority alerting signal, and the display on the telephone shows “rt”, which indicates
a returned call from a failed transfer operation.
Transfer upon hang-up
Provides you with the ability to transfer a call by hanging up instead of having to press the
transfer button a second time. You would press the transfer button, dial the number the call is
being transferred to and then hang up. This is an optional feature at the system level. You will
still be able to transfer a call by pressing the transfer button a second time.
Trunk-to-trunk transfer
Allows the attendant or telephone user to connect an incoming trunk call to an outgoing trunk
call. This feature is particularly useful when a caller outside the system calls a user or attendant
and requests a transfer to another outside number. For example, a worker, away on business,
can call in and have the call transferred elsewhere. The system assures that incoming central
office (CO) trunks without disconnect supervision are not transferred to outgoing trunks or other
incoming central office trunks without disconnect supervision.
Issue 6 May 2009
265
Telephony
Trunk flash
Trunk flash allows a feature or function button on a multifunction telephone or attendant console
to be assigned as a flash button. Pressing this button while connected to a trunk (which must
have been administered to allow trunk flash) causes the system to send a flash signal out over
the connected trunk.
Trunk flash enables multifunction telephones to access central office customized services that
are provided by the central office to which the system running Avaya AuraTM Communication
Manager is connected. These services are electronic features, such as conference and
transfer, that are accessed by a sequence of flash signal and dial signals from the system
station on an active trunk call.
The trunk flash feature can help to reduce the number of trunk lines connected to the system.
“Digit 1 as flash” as used in Italy, and the United Kingdom will not serve as the flash button in
this application.
266 Avaya AuraTM Communication Manager Overview
Index
Index
Numerical
2B-channel transfer . . . . . . . . . . . . . . . . 67
800-service trunks . . . . . . . . . . . . . . . . . 144
802.1p/Q . . . . . . . . . . . . . . . . . . . . . 154
A
AAA, see Authentication, Authorization, and Accounting
Services (AAA)
AAR, see Automatic Alternate Routing (AAR)
AAR/ARS, see Automatic Alternate Routing/Automatic
Route Selection (AAR/ARS)
AAS, see Auto-Available Split (AAS)
abandoned call . . . . . . . . . . . . . . . . . . 51
abandoned call search . . . . . . . . . . . . . . . 51
Abbreviated Dialing (AD) . . . . . . . . . . . . . . 235
labeling . . . . . . . . . . . . . . . . . . . . 235
on-hook programming . . . . . . . . . . . . . . 235
ABCD tone support . . . . . . . . . . . . . . . . 236
abort conference on hangup . . . . . . . . . . . . 73
abort transfer . . . . . . . . . . . . . . . . . . . 265
ACB, see Automatic Call Back (ACB)
access security gateway (ASG) . . . . . . . . . . . 197
accessing the attendant . . . . . . . . . . . . . . 37
ACD, see Automatic Call Distribution (ACD)
active dialing . . . . . . . . . . . . . . . . . . . 236
ACTR, see Automatic Customer Telephone Rearrangement
(ACTR)
ACW, see After Call Work (ACW)
AD, see Abbreviated Dialing (AD)
add/remove skills . . . . . . . . . . . . . . . . . 64
adjunct route support for network call redirection . . . 47
adjunct routing . . . . . . . . . . . . . . . . . . . 52
Adjunct Switch Application Interface (ASAI) . . . . . 34
adjunct route support for network call redirection . 47
co-resident DEFNINTY LAN Gateway (DLG) . . . 48
Direct Agent Announcement (DAA) . . . . . . . 48
flexible billing . . . . . . . . . . . . . . . . . . 48
pending work mode change . . . . . . . . . . . 49
trunk group identification . . . . . . . . . . . . 49
User-to-User Information (UUI) . . . . . . . . . 49
administered connections . . . . . . . . . . . . . 175
administrable language displays . . . . . . . . . . 95
administrable loss plan . . . . . . . . . . . . . . . 95
administrable time-out on call timer . . . . . . . . . 236
administration
automatic routing . . . . . . . . . . . . . . . . 181
call management . . . . . . . . . . . . . . . . 54
change notification . . . . . . . . . . . . . . . 213
duplicate agent login ID . . . . . . . . . . . . . 64
agent-loginID skill pair increase . . . . . . . . 64
monitoring calls . . . . . . . . . . . . . . . . . 44
property management . . . . . . . . . . . . . . 92
site . . . . . . . . . . . . . . . . . . . . . . 216
Administration Without Hardware (AWOH) 113, 117, 119,
213
admission control . . . . . . . . . . . . . . . . . 157
Advanced Encryption Standard (AES) . . . . . 159, 200
encryption algorithm for bearer channels . . . . 200
Advanced Private Line Termination (APLT) . . . . . 132
advanced vector routing. . . . . . . . . . . . . . . 59
Advice of Charge (AOC) . . . . . . . . . . . . . 218
AE Services, see Application Enablement Services (AE
Services)
AEA, see Avaya Encryption Algorithm (AEA)
AES, see Advanced Encryption Standard (AES)
After Call Work (ACW) . . . . . . . . . . . . . . . 62
ALI, see Automatic Location Information (ALI)
alphanumeric dialing . . . . . . . . . . . . . . 89, 236
alternate facility restriction levels . . . . . 181, 198, 213
alternate gatekeeper and registration addresses . . 150
alternate operations support system alarm . . . . . 198
analog
CAMA - E911 trunk group . . . . . . . . . . . 209
TTY over analog trunks . . . . . . . . . . . . 162
ANI, see Automatic Number Identification (ANI)
announcements . . . . . . . . . . . . . . . . . 214
increased support . . . . . . . . . . . . . . . 216
multiple music sources. . . . . . . . . . . . . 229
Voice Announcement over LAN (VAL) Manager . 216
announcements for precedence calling . . . . . . 251
answer detection . . . . . . . . . . . . . . . . . 185
answer supervision by time-out . . . . . . . . . . 185
AOC, see Advice of Charge (AOC)
API, see Application Programming Interface (API)
APLT, see Advanced Private Line Termination (APLT)
Application Enablement Services (AE Services) . . . 31
bundled server option . . . . . . . . . . . . . . 32
software-only option . . . . . . . . . . . . . . . 31
Application Programming Interface (API) . . . . . . . 31
Adjunct Switch Application Interface (ASAI) . . . . 34
device and media control API . . . . . . . . . . 33
JTAPI . . . . . . . . . . . . . . . . . . . . . 34
TSAPI . . . . . . . . . . . . . . . . . . . . . 34
Application Server Interface (ASI) . . . . . . . . 78, 177
approximate charge for calls . . . . . . . . . . . 218
ARS, see Automatic Route Selection (ARS)
ASA, see Average Speed of Answer (ASA) routing
ASAI, see Adjunct Switch Application Interface (ASAI)
ASCA, see Avaya Software Compatibility Audit (ASCA) tool
ASCII character set . . . . . . . . . . . . . . . . . 92
ASI, see Application Server Interface (ASI)
Issue 6 May 2009
267
Index
asynchronous links. . . . . . . . . . . . . . 176, 228
Asynchronous Transfer Mode (ATM) . . . . . 121, 127
Circuit Emulation Service (ATM-CES) . . . . . . 127
Port Network Connectivity (ATM-PNC) . . . . . . 121
over WAN . . . . . . . . . . . . . . . . . . 121
ATM-CES, see Asynchronous Transfer Mode (ATM), Circuit
Emulation Service (ATM-CES)
ATM-PNC, see Asynchronous Transfer Mode (ATM), Port
Network Connectivity (PNC)
attendant
auto manual splitting . . . . . . . . . . . . . . 44
auto start . . . . . . . . . . . . . . . . . . . . 44
automated . . . . . . . . . . . . . . . . . . . 40
backup . . . . . . . . . . . . . . . . . . . . . 38
backup alerting . . . . . . . . . . . . . . . . . 40
call handling . . . . . . . . . . . . . . . . . . 39
call waiting . . . . . . . . . . . . . . . . . . . 40
calling of inward restricted stations . . . . . . . . 40
conferencing . . . . . . . . . . . . . . . . . . 41
control of trunk group access . . . . . . . . . . 44
crisis alert . . . . . . . . . . . . . . . . . . . 45
dial access to . . . . . . . . . . . . . . . . . . 37
direct extension selection . . . . . . . . . . . . 45
display of Class of Restriction (COR) . . . . . . . 167
do not split . . . . . . . . . . . . . . . . . . . 44
functions using Distributed Communications System
(DCS) protocol . . . . . . . . . . . . . . 38, 147
control of trunk group access . . . . . . . . . 38
direct trunk group selection . . . . . . 39, 45, 147
display . . . . . . . . . . . . . . . . . 43, 147
inter-PBX attendant calls . . . . . . . . . . . 39
increased consoles . . . . . . . . . . . . . . . 43
individual access to . . . . . . . . . . . . . . . 37
intrusion . . . . . . . . . . . . . . . . . . . . 39
listed directory number . . . . . . . . . . . . . 41
lockout - privacy . . . . . . . . . . . . . . . . 39
override of diversion features . . . . . . . . . . 41
position report . . . . . . . . . . . . . . . . . 225
priority queue . . . . . . . . . . . . . . . . . . 42
QSIG Centralized Attendant Service (CAS) . . . . 167
recall . . . . . . . . . . . . . . . . . . . . . . 37
release loop operation . . . . . . . . . . . . . 42
return call . . . . . . . . . . . . . . . . . . . 167
room status . . . . . . . . . . . . . . . . . 38, 89
serial calling . . . . . . . . . . . . . . . . . . 42
split swap . . . . . . . . . . . . . . . . . . . 39
timed reminder . . . . . . . . . . . . . . . . . 42
trunk group busy/warning indicators . . . . . . . 46
trunk identification . . . . . . . . . . . . . . . 46
vectoring . . . . . . . . . . . . . . . . . . . . 40
Visually Impaired Attendant Service (VIAS) . . . . 46
audible message waiting . . . . . . . . . . . . . . 107
AUDIX
record on messaging . . . . . . . . . . . . . . 106
Authentication, Authorization, and Accounting Services
(AAA) . . . . . . . . . . . . . . . . . . . . . . 197
268 Avaya AuraTM Communication Manager Overview
authorization codes - 13 digits . . . . . . . . . 198, 214
auto answer ICOM . . . . . . . . . . . . . . . . 248
auto callback - QSIG call completion . . . . . . . . 164
auto fallback to primary for H.248 gateways . . . . 187
auto manual splitting . . . . . . . . . . . . . . . . 44
auto reserve agents . . . . . . . . . . . . . . . . 55
auto start. . . . . . . . . . . . . . . . . . . . . . 44
Auto-Available Split (AAS) . . . . . . . . . . . . . 52
auto-in work mode . . . . . . . . . . . . . . . . . 52
automated attendant . . . . . . . . . . . . . . . . 40
automatic alternate conditional routing . . . . . . . 148
Automatic Alternate Routing (AAR) . . . . . . . . 181
Automatic Alternate Routing/Automatic Route Selection
(AAR/ARS) . . . . . . . . . . . . . . . . . . . . 78
dialing without FAC . . . . . . . . . . . . . . 182
overlap sending . . . . . . . . . . . . . . . . 182
partitioning . . . . . . . . . . . . . . . . 182, 183
automatic answer
intercom . . . . . . . . . . . . . . . . . . . 248
internal . . . . . . . . . . . . . . . . . . . . 248
Automatic Call Back (ACB) . . . . . . . . . . . . 237
for analog telephones . . . . . . . . . . . . . 237
Automatic Call Distribution (ACD) . . . . 50, 52, 57, 239
automatic circuit assurance . . . . . . . . 147, 214, 222
Automatic Customer Telephone Rearrangement (ACTR)113
automatic hold . . . . . . . . . . . . . . . . . . 237
Automatic Location Information (ALI). . . . . . 116, 117
Automatic Number Identification (ANI) . . . . . . 49, 52
Incoming Automatic Number Identification . . . . 52
Outgoing Automatic Number Identification . . . . 53
Automatic Route Selection (ARS) . . . . . . . . . 182
automatic routing features . . . . . . . . . . . . . 181
automatic selection of Direct Inward Dialing (DID) numbers
89
automatic transmission measurement system . . . 214
automatic wakeup . . . . . . . . . . . . . . . . . 89
auxiliary trunks . . . . . . . . . . . . . . . . . . 132
Avaya business advocate . . . . . . . . . . . . . . 55
enhancements
auto reserve agents . . . . . . . . . . . . . 55
call selection override per skill . . . . . . . . . 55
dynamic percentage adjustment . . . . . . . . 55
dynamic queue position . . . . . . . . . . . . 55
dynamic threshold adjustment . . . . . . . . . 55
Least Occupied Agent (LOA) . . . . . . . . . 65
logged-in advocate agent counting . . . . . . 55
percent allocation distribution . . . . . . . . . 56
reserve agent time in queue activation . . . . . 56
VuStats . . . . . . . . . . . . . . . . . . . . . 71
Avaya call center
basic . . . . . . . . . . . . . . . . . . . . . . 56
deluxe . . . . . . . . . . . . . . . . . . . . . 56
elite . . . . . . . . . . . . . . . . . . . . . . 56
features supported on the Avaya G700 Media Gateway
56
Avaya Call Management System (CMS) . . . . . 57, 231
Index
dual links to CMS . . . . . . . . . . . . . . . . 63
measurement of ATM . . . . . . . . . . . . 63, 127
site statistics for remote port networks . . . . . . 70
Avaya computer telephony . . . . . . . . . . . . . 47
Avaya Directory Enabled Management (DEM) . . . . 215
Avaya Encryption Algorithm (AEA) . . . . . . 159, 200
Avaya Extension to Cellular. . . . . . . . . . . . . 114
off-PBX station (OPS) . . . . . . . . . . . . . . 115
Avaya Installation Wizard . . . . . . . . . . . . . . 26
Avaya Integrated Management . . . . . . . . 215, 216
Avaya IP agent . . . . . . . . . . . . . . . . . . 83
Avaya IP Softphone . . . . . . . . . . . . . . . . 83
for pocket PC . . . . . . . . . . . . . . . . . . 84
Avaya one-X Communicator . . . . . . . . . . . . 85
Avaya one-X Portal as software-only phone . . . . . 85
Avaya SIP softphone . . . . . . . . . . . . . . . . 86
Avaya site administration . . . . . . . . . . . . . . 216
Avaya SoftConsole. . . . . . . . . . . . . . . . . 86
RoadWarrior mode . . . . . . . . . . . . . . . 86
Telecommuter mode . . . . . . . . . . . . . . 87
Avaya Software Compatibility Audit (ASCA) tool . . . 225
Avaya video telephony solution . . . . . . . . . . . 237
Avaya virtual routing . . . . . . . . . . . . . . . . 57
Avaya VoIP Monitoring Manager (VMON) . . . 147, 217
Avaya Wireless Telephone Solutions (AWTS) . . . . 114
Average Speed of Answer (ASA) routing . . . . . . 59
AWOH, see Administration Without Hardware (AWOH)
AWTS, see Avaya Wireless Telephone Solutions (AWTS)
B
backup alerting . . . . . . . . . . . . . . . . 40, 207
barrier codes . . . . . . . . . . . . . . . . 207, 217
Basic Call Management System (BCMS) . . . . . . 54
reports . . . . . . . . . . . . . . . . . . . . . 54
Basic Rate Interface (BRI) . . . . . . . . . . . . . 135
BCD, see Binary Coded Decimal (BCD)
BCMS, see Basic Call Management System (BCMS)
Bellcore calling name ID . . . . . . . . . 95, 143, 238
Best Service Routing (BSR) . . . . . . . . . . . . 59
polling over IP without B channel . . . . . . . . 59
Binary Coded Decimal (BCD) . . . . . . . . . . . . 92
block CMS Move Agent events . . . . . . . . . . . 49
block collect call . . . . . . . . . . . . . . . . 96, 199
blockage study report . . . . . . . . . . . . . . . 225
BRI, see Basic Rate Interface (BRI)
bridged call appearance
multi-appearance telephone . . . . . . . . . . . 238
single-line telephone . . . . . . . . . . . . . . 239
BSR, see Best Service Routing (BSR)
bulletin board . . . . . . . . . . . . . . . . 104, 217
busy tone disconnect . . . . . . . . . . . . . . . . 96
busy verification of telephones and trunks . . . . . . 218
C
CAC, see Call Admission Control (CAC)
CAG, see Coverage Answer Group (CAG)
Cajun rules . . . . . . . . . . . . . . . . . . . . 155
call accounting
Xiox . . . . . . . . . . . . . . . . . . . . . . 93
Call Admission Control (CAC) bandwidth management155
Call Admission Control using Bandwidth Limits (CAC-BL)
123
call-by-call service selection . . . . . . . . . . . . 135
call center . . . . . . . . . . . . . . . . . . . . . 47
messaging . . . . . . . . . . . . . . . . . . . 60
release control . . . . . . . . . . . . . . . . . 58
call charge information . . . . . . . . . . . . . . 218
call-classifier board . . . . . . . . . . . . . . . . 185
call completion . . . . . . . . . . . . . . . . . . 164
call control . . . . . . . . . . . . . . . . . . . . 156
call coverage . . . . . . . . . . . . . . . 111, 170, 239
alphanumeric field designation . . . . . . . . . 239
and CAS . . . . . . . . . . . . . . . . . . . 170
changeable coverage paths . . . . . . . . . . 239
enhanced coverage and ringback for logged off IP/PSA/
TTI terminals . . . . . . . . . . . . . . . . 240
redirection intervals . . . . . . . . . . . . . . 242
report . . . . . . . . . . . . . . . . . . . . . 225
time of day . . . . . . . . . . . . . . . . . . 240
Call Coverage Remote Off Net (C-CRON) . . . . . 170
Call Detail Recording (CDR) . . . . . . 78, 176, 177, 219
display of physical extension . . . . . . . . . . 219
call distribution based on skill . . . . . . . . . . . . 65
call forwarding
all calls . . . . . . . . . . . . . . . . . . . . 241
busy/do not answer . . . . . . . . . . . . . . 240
diversion . . . . . . . . . . . . . . . . . . . 165
override . . . . . . . . . . . . . . . . . . . . 241
call handling . . . . . . . . . . . . . . . . . . . . 39
Call Independent Signaling Connections (CISC) . . 166
Call Management System (CMS)
measurement of ATM . . . . . . . . . . . . 63, 127
call offer . . . . . . . . . . . . . . . . . . . . . 166
Call Offer, see attendant, intrusion
call park . . . . . . . . . . . . . . . . . . . . . 242
call pickup . . . . . . . . . . . . . . . . . . . . 243
group call pickup . . . . . . . . . . . . . . . 243
call prompting . . . . . . . . . . . . . . . . . . . 58
call center messaging . . . . . . . . . . . . . . 60
data collection . . . . . . . . . . . . . . . . . . 58
Data In/Voice Answer (DIVA). . . . . . . . . . . 58
call redirection intervals . . . . . . . . . . . . . . 242
call redirection to multimedia endpoint . . . . . . . . 79
call restrictions . . . . . . . . . . . . . . . . 199, 220
call routing . . . . . . . . . . . . . . . . . . . . 181
call selection override per skill . . . . . . . . . . . . 55
call transfer . . . . . . . . . . . . . . . . . . . 166
call vectoring . . . . . . . . . . . . . . . . . . . . 59
advanced vector routing . . . . . . . . . . . . . 59
Average Speed of Answer (ASA) routing . . . . . 59
Class of Restriction (COR) for VDN . . . . . . . . 61
Issue 6 May 2009
269
Index
Expected Wait Time (EWT) . . . . . . . . . . . 60
holiday vectoring . . . . . . . . . . . . . . . . 60
call waiting . . . . . . . . . . . . . . . . . . . . 40
Call Work Codes (CWC) . . . . . . . . . . . . . . 62
called name ID . . . . . . . . . . . . . . . . . . 166
Caller Emergency Service Identification (CESID) . . . 134
Caller ID (ICLID)
on analog trunks . . . . . . . . . . . . . 143, 243
on digital trunks . . . . . . . . . . . . . . 143, 243
Caller Information Forwarding (CINFO) . . . . . . . 62
calling of inward restricted stations . . . . . . . . . 40
calling party/billing number . . . . . . . . . . . . . 220
calls
charging for service . . . . . . . . . . . . . . . 218
disconnecting. . . . . . . . . . . . . . . . . . 244
monitoring . . . . . . . . . . . . . . . . . . . 44
placing . . . . . . . . . . . . . . . . . . . . . 44
redirecting . . . . . . . . . . . . . . . . . . . 240
routing capabilities . . . . . . . . . . . . . . . 183
CAMA, see Centralized Automatic Message Accounting
(CAMA)
camp-on/busy-out . . . . . . . . . . . . . . . . . 154
capacities . . . . . . . . . . . . . . . . . . . . . 26
CAS, see Centralized Attendant Service (CAS)
CCMS, see Control Channel Message Set (CCMS)
C-CRON, see Call Coverage Remote Off Net (C-CRON)
CCSA, see Common Control Switching Arrangements
(CCSA)
CDR, see Call Detail Recording (CDR)
Center Stage Switch (CSS) . . . . . . . . . . 121, 122
separation of . . . . . . . . . . . . . . . . . . 122
Central Office (CO) . . . . . . . . . . . . . 133, 144
support on G700 Media Gateway - Russia . . 97, 133
Centralized Attendant Service (CAS) . . . . . . 43, 167
Centralized Automatic Message Accounting (CAMA) 134,
209
Centralized voice mail (Tenovis) . . . . . . . . . . 108
CES, see Circuit Emulation Service (CES)
CESID, see Caller Emergency Service Identification
(CESID)
changeable coverage paths . . . . . . . . . . . . 239
check-in/check-out . . . . . . . . . . . . . . . . . 90
CIDR, see Classless Interdomain Routing (CIDR)
CINFO see Caller Information Forwarding (CINFO)
Circuit Emulation Service (CES) . . . . . . . . . . 127
circuit switched . . . . . . . . . . . . . . . 122, 127
circular station hunt group . . . . . . . . . . . 62, 243
CISC, see Call Independent Signaling Connections (CISC)
CLAN, see Control LAN (CLAN)
Class of Restriction (COR) . 61, 140, 167, 199, 220, 264
attendant display . . . . . . . . . . . . . . . . 167
for VDN . . . . . . . . . . . . . . . . . . . . 61
Class of Service (COS) . . . . . . . . . . . . 220, 230
Classless Interdomain Routing (CIDR) . . . . 151, 221
CMS, see Call Management System (CMS)
CO, see Central Office (CO)
270 Avaya AuraTM Communication Manager Overview
code calling access . . . . . . . . . . . . . . . . . 79
codecs . . . . . . . . . . . . . . . . . . . . . . 155
Common Control Switching Arrangements (CCSA) . 132
communication device support . . . . . . . . . . . 83
Communication Manager . . . . . . . . . . . . 25, 215
fault/performance manager . . . . . . . . . . 216
Octel QSIG integration . . . . . . . . . . . . . 167
overview . . . . . . . . . . . . . . . . . . . . 25
PC console . . . . . . . . . . . . . . . . . . . 85
Communication Manager Messaging . . . . . . . 103
Completion of Calls on No Reply (CCNR) . . . . . 136
Completion of Calls to Busy Subscriber (CCBS) . . 136
Complex private numbering plan support . . . . . . 167
Computer Telephony Integration (CTI) . . . . . . 47, 62
concurrent user sessions . . . . . . . . . . . . . 222
conference/transfer display prompts . . . . . . . . . 74
conference/transfer toggle/swap . . . . . . . . . . . 74
conferencing . . . . . . . . . . . . . . . . . . 73, 244
abort conference on hangup . . . . . . . . . . . 73
automatic answer intercom . . . . . . . . . . . . 80
automatic intercom . . . . . . . . . . . . . . . 80
code calling access . . . . . . . . . . . . . . . 79
conference/transfer display prompts . . . . . . . 74
dial intercom . . . . . . . . . . . . . . . . . . 80
expanded meet-me . . . . . . . . . . . . . . . 75
group listen . . . . . . . . . . . . . . . . . . . 74
group paging . . . . . . . . . . . . . . . . . . 80
hold/unhold . . . . . . . . . . . . . . . . . . . 74
loudspeaker paging access . . . . . . . . . . . 81
manual signaling . . . . . . . . . . . . . . . . 81
meet-me . . . . . . . . . . . . . . . . . . . . 75
multimedia . . . . . . . . . . . . . . . . . 79, 179
no dial tone . . . . . . . . . . . . . . . . . . . 75
no hold conference . . . . . . . . . . . . . . . 75
select line appearance . . . . . . . . . . . . . . 76
selective party display and drop . . . . . . . . . 76
six party . . . . . . . . . . . . . . . . . . . . 73
three party . . . . . . . . . . . . . . . . . . . 73
transfer toggle/swap . . . . . . . . . . . . . . . 74
whisper page . . . . . . . . . . . . . . . . . . 81
with attendant . . . . . . . . . . . . . . . . . . 41
configuration manager . . . . . . . . . . . . . . 215
Connected Party Number (CPN) . . . . . . . . . 49, 207
restriction per call . . . . . . . . . . . . . . . 207
restriction per line . . . . . . . . . . . . . . . 208
connection preserving failover/failback for H.248 media
gateways . . . . . . . . . . . . . . . . . . . . 188
Connection Preserving Migration (CPM) . . . . . . 188
connection preserving upgrades for duplex servers . 188
console
Avaya SoftConsole . . . . . . . . . . . . . 86, 87
PC . . . . . . . . . . . . . . . . . . . . . . . 85
consult . . . . . . . . . . . . . . . . . . . . . . 244
Control Channel Message Set (CCMS) . . . . . . 123
Control LAN (CLAN)
load balancing . . . . . . . . . . . . . . . . 155
Index
multiple network regions . . . . . . . . . 151, 193
control of trunk group access . . . . . . . . . . . . 38
COR, see Class of Restriction (COR)
Co-residency with SIP . . . . . . . . . . . . . . . 29
co-resident DEFNINTY LAN Gateway (DLG) . . . . 48
COS, see Class of Service (COS)
Coverage Answer Group (CAG) . . . . . . . . . . 239
coverage callback . . . . . . . . . . . . . . . . . 244
coverage incoming call identification . . . . . . . . 244
coverage points report . . . . . . . . . . . . . . . 225
CPM, see Connection Preserving Migration (CPM)
CPN, see Connected Party Number (CPN)
crisis alert
to a digital numeric pager . . . . . . . . . . . . 208
to a digital station . . . . . . . . . . . . . . . . 208
to an attendant console . . . . . . . . . . . 45, 209
CSS, see Center Stage Switch (CSS)
CTI, see Computer Telephony Integration (CTI)
customer selection of VIP DID numbers . . . . . . . 90
Customer Telephone Activation . . . . . . . . . . . 222
customer-provided equipment alarm . . . . . 199, 222
CVLAN . . . . . . . . . . . . . . . . . . . . . . 32
CWC, see Call Work Codes (CWC)
D
DAA, see Direct Agent Announcement (DAA)
daily wakeup . . . . . . . . . . . . . . . . . . . 90
data call setup . . . . . . . . . . . . . . . . . . . 175
data calls . . . . . . . . . . . . . . . . . . . . . 89
data collection . . . . . . . . . . . . . . . . . . . 58
data conference . . . . . . . . . . . . 77, 79, 177, 179
data conferencing (T.120) via ESM . . . . . . . 79, 179
data hot line . . . . . . . . . . . . . . . . . . . . 175
Data In/Voice Answer (DIVA) . . . . . . . . . . . . 58
data interfaces . . . . . . . . . . . . . . . . . . . 175
administered connections . . . . . . . . . . . . 175
data call setup . . . . . . . . . . . . . . . . . 175
data hot line . . . . . . . . . . . . . . . . . . 175
data privacy . . . . . . . . . . . . . . . . . . 175
data restriction . . . . . . . . . . . . . . . . . 176
default dialing . . . . . . . . . . . . . . . . . 176
multimedia
Application Server Interface (ASI). . . . . . . 177
call early answer on vectors and stations . . . 177
call redirection to multimedia endpoint . . . . 179
calling . . . . . . . . . . . . . . . . . . . 177
hold, conference, transfer, and drop . . . . . 179
Multimedia Call Handling (MMCH) . . . . . . 178
multiple-port networks . . . . . . . . . . . . 179
pass advice of charge information to world class BRI
endpoints . . . . . . . . . . . . . . . . . . . 179
data privacy . . . . . . . . . . . . . . . . . 175, 199
data restriction . . . . . . . . . . . . . . . . 176, 200
Daylight Savings Time rules change . . . . . . . . 151
DCS+, see Distributed Communications System plus
(DCS+)
DCS, see Distributed Communications System (DCS)
protocol
DDC, see Direct Department Calling (DDC)
default dialing . . . . . . . . . . . . . . . . . . 176
defense switched network (DSN) . . . . . . . . . 251
DEFINITY LAN Gateway (DLG) . . . . . . . . . . . 34
DEFINITY Wireless Business System (DWBS) . . . . 114
deluxe paging . . . . . . . . . . . . . . . . . . . 81
DEM, see Avaya Directory Enabled Management (DEM)
destination voice endpoint . . . . . . . . . . . . 78, 178
device and media control API . . . . . . . . . . . . 33
dial access to attendant . . . . . . . . . . . . . . . 37
Dial Plan Expansion (DPE) . . . . . . . . . . 125, 171
Dial Plan Transparency for LSP and ESS . . . . . 189
dial-by-name . . . . . . . . . . . . . . . . . . . . 90
Dialed Number Identification Service (DNIS) . . . . . 63
DID, see Direct Inward Dialing (DID)
differentiated services (DiffServ) . . . . . . . . . . 155
type-of-service value . . . . . . . . . . . . . 149
Digital Communications Protocol (DCP) . . . . . . . 99
digital interfaces . . . . . . . . . . . . . . . . . 135
Digital Service 1 (DS1) trunks . . . . . . . 127, 143
E1 . . . . . . . . . . . . . . . . . . . . 128, 143
T1 . . . . . . . . . . . . . . . . . . . . 128, 144
TTY over digital trunks . . . . . . . . . . . . . 162
digital multiplexed interface . . . . . . . . . . . . 133
bit-oriented signalling . . . . . . . . . . . . . 133
message-oriented signalling . . . . . . . . . . 133
Digital Service 1 (DS1) trunks . . . . . . . 127, 143, 144
digital telephones
2420 DCP
voice mail retrieval button . . . . . . . . . . . 110
DIOD, see Direct Inward/Outward Dialing (DIOD)
Direct Agent Announcement (DAA) . . . . . . . . . 48
direct agent calling . . . . . . . . . . . . . . . . . 63
Direct Department Calling (DDC) . . . . . . . . . 239
direct extension selection . . . . . . . . . . . . . . 45
direct extension selectors (DXS) . . . . . . . . . . . 43
Direct Inward Dialing (DID) . . . . . . . . . . 134, 144
automatic number selection . . . . . . . . . . . 89
Direct Inward/Outward Dialing (DIOD) . . . . . 134, 145
direct trunk group selection . . . . . . . . . 39, 45, 147
directory . . . . . . . . . . . . . . . . . . . . . 240
disconnecting unanswered calls . . . . . . . . . . 244
display . . . . . . . . . . . . . . . . . . . . . 43, 147
and drop conferencing . . . . . . . . . . . . . . 76
ARP report . . . . . . . . . . . . . . . . . . 225
VDN for route-to DAC . . . . . . . . . . . . . . 61
distinctive alerting . . . . . . . . . . . . . . . . 170
distinctive ringing . . . . . . . . . . . . . . . . . 245
maintain external ring tone after internal transfer 245
Distributed Communications System (DCS) protocol 147,
164, 171, 245, 249
attendant functions . . . . . . . . . . . . . 38, 147
control of trunk group access . . . . . . . . . 38
direct trunk group selection . . . . . . . . 39, 147
Issue 6 May 2009
271
Index
display . . . . . . . . . . . . . . . . . 43, 147
inter-PBX attendant calls . . . . . . . . . . . 39
automatic circuit assurance . . . . . . . . 147, 222
Italy . . . . . . . . . . . . . . . . . . . . 96, 148
over ISDN-PRI D-channel . . . . . . . . . . . . 148
with reroute . . . . . . . . . . . . . . . . . . 148
Distributed Communications System plus (DCS+) . . 148
DIVA, see Data In/Voice Answer (DIVA)
DLG, see co-resident DEFNINTY LAN Gateway (DLG)
DLG, see DEFINITY LAN Gateway (DLG)
DNIS, see Dialed Number Identification Service (DNIS)
do not answer reason code . . . . . . . . . . . . . 117
do not disturb . . . . . . . . . . . . . . . . . . . 91
do not split . . . . . . . . . . . . . . . . . . . . 44
DPE, see Dial Plan Expansion (DPE)
DS1, see Digital Service 1 (DS1)
DSN, see defense switched network (DSN)
dual homing . . . . . . . . . . . . . . . . . . . . 252
dual links to CMS . . . . . . . . . . . . . . . . . 63
dual wakeup . . . . . . . . . . . . . . . . . . . . 91
duplicate agent login ID administration . . . . . . . 64
agent-loginID skill pair increase . . . . . . . . . 64
DWBS, see DEFINITY Wireless Business System (DWBS)
DXS, see direct extension selectors (DXS)
dynamic jitter buffers . . . . . . . . . . . . . . . . 155
dynamic percentage adjustment . . . . . . . . . . 55
dynamic queue position . . . . . . . . . . . . . . 55
dynamic threshold adjustment . . . . . . . . . . . 55
E
E&M signaling - continuous and pulsed . . . . . 98, 134
E1 digital interface . . . . . . . . . . . . . . 128, 143
E911 . . . . . . . . . . . . . . . . . . . . 134, 209
E911 device location for IP telephones . . . . . . . 117
E911 ELIN for IP wired extensions . . . . . . . . . 116
EAS, see Expert Agent Selection (EAS)
EC500, see Avaya Extension to Cellular
echo cancellation circuit pack . . . . . . . . . 128, 143
ECS, see Enterprise Communication Server (ECS)
ECT, see ETSI Explicit Call Transfer (ECT) signaling
EIW, see Electronic pre-Installation Worksheet (EIW)
Electronic pre-Installation Worksheet (EIW) . . . . . 27
Electronic Tandem Network (ETN) . . . . . . . . . 148
automatic alternate conditional routing . . . . . . 148
extension number portability . . . . . . . . . . . 149
traveling class marks . . . . . . . . . . . . . . 185
trunk signaling and error recovery . . . . . . . . 149
emergency access to the attendant . . . . . . . . . 209
emergency and journal report . . . . . . . . . . . . 225
emergency calls . . . . . . . . . . . . . 116, 134, 209
emergency calls from unnamed IP endpoints . . . . 245
emergency transfer . . . . . . . . . . . . . . . . 193
EMU, see Enterprise Mobility User (EMU)
Enbloc Dialing and Call Type Digit Analysis . . . . . 183
encryption algorithm for bearer channels . . . . . . 200
272 Avaya AuraTM Communication Manager Overview
end office access line hunting . . . . . . . . . . . 252
enhanced abbreviated dialing . . . . . . . . . . . 246
enhanced coverage and ringback for logged off IP/PSA/TTI
terminals . . . . . . . . . . . . . . . . . . . . 240
enhanced information forwarding . . . . . . . . . . 57
enhanced logging of user actions . . . . . . . . . 227
enhanced night service . . . . . . . . . . . . . . 254
Enhanced Private Switched Communications Service
(EPSCS) . . . . . . . . . . . . . . . . . . . . 132
enhanced security logging . . . . . . . . . . . . 201
Enhanced Software License Program (ESLP) . . . 248
enhanced telephone display . . . . . . . . . . . . 246
Enterprise Communication Server (ECS) . . . . . . . 98
Enterprise Mobility User (EMU) . . . . . . . . . . 247
enhancements . . . . . . . . . . . . . . . . 247
Enterprise Survivable Servers (ESS) . . . . . . . . 189
Enterprise Wide Licensing (EWL) . . . . . . . . . 248
EPN, see Expansion Port Network (EPN)
EPSCS, see Enhanced Private Switched Communications
Service (EPSCS)
ESLP, see Enhanced Software License Program (ESLP)
ESM, see Expansion Services Module (ESM)
ESS, see Enterprise Survivable Servers (ESS)
ETN, see Electronic Tandem Network (ETN)
ETSI Explicit Call Transfer (ECT) signaling . . . . . . 67
ETSI functionality . . . . . . . . . . . . . . . . . 135
ETSI Completion of Calls to Busy Subscriber (CCBS)
and on No Reply (CCNR) . . . . . . . . . . . 136
European Union . . . . . . . . . . . . . . . . . 121
EWL, see Enterprise Wide Licensing (EWL)
EWT, see Expected Wait Time (EWT)
expanded meet-me conferencing . . . . . . . . . . 75
Expansion Port Network (EPN) . . . . . . . . . . 121
Expansion Services Module (ESM) . . 77, 79, 177, 179
Expected Wait Time (EWT) . . . . . . . . . . . . . 60
Expert Agent Selection (EAS) . . . . . . . . 48, 50, 64
add/remove skills . . . . . . . . . . . . . . . . 64
call distribution based on skill . . . . . . . . . . 65
queue to best ISDN support . . . . . . . . . . . 65
extended trunk access . . . . . . . . . . . . . . 173
extension number portability . . . . . . . . . . . . 149
external device alarming . . . . . . . . . . . . . 223
F
FAC, see Feature Access Code (FAC)
facility and non-facility associated signaling . . . . 136
facility busy indication . . . . . . . . . . . . . . . 223
facility restriction levels and traveling class marks201, 223
facility test calls. . . . . . . . . . . . . . . . . . 223
far end mute, see selective conference mute
faxes, sending and receiving over IP . . . . . . . . 157
Feature Access Code (FAC) . . . . . . . . . . . 165
feature plus . . . . . . . . . . . . . . . . . . . 136
FIFO, see First In/First Out (FIFO)
firmware download . . . . . . . . . . . . . . . . 224
First In/First Out (FIFO) . . . . . . . . . . . . . . . 57
Index
five EPN maximum in MCC1 Media Gateways . . . . 224
flexible billing . . . . . . . . . . . . . . . . . 48, 144
Foreign Exchange (FX) . . . . . . . . . . . . 134, 145
FX, see Foreign Exchange (FX)
G
Gateway Installation Wizard (GIW) . . . . .
Generalized Conference Call (GCC) . . . .
generalized route selection . . . . . . . . .
GIW, see Gateway Installation Wizard (GIW)
go to cover . . . . . . . . . . . . . . . .
group call pickup . . . . . . . . . . . . . .
group listen . . . . . . . . . . . . . . . .
group paging . . . . . . . . . . . . . . .
. . . . 28
. . 79, 179
. . . . 183
.
.
.
.
.
.
.
.
.
.
.
.
. 248
. 243
. 74
. 80
H
H.248 link encryption . . . . . . . .
H.248 media gateway control . . . .
H.323 trunk . . . . . . . . . . . .
hairpinning . . . . . . . . . . . .
historical reports . . . . . . . . . .
history report
parsing capabilities . . . . . . .
hold . . . . . . . . . . . . . . . .
hold, conference, transfer, and drop .
hold/unhold conference . . . . . .
holiday vectoring . . . . . . . . . .
hospitality . . . . . . . . . . . . .
hot line service . . . . . . . . . .
housekeeping status . . . . . . . .
HP DL380G2 server, support for . .
hunt group measurements report . .
hunt groups . . . . . . . . . . . .
circular station . . . . . . . . .
circular station hunting . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
. 201
. 122
. 129
. 160
. 54
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
. 227
. 248
. 79
. 74
. 60
. 89
. 249
. 91
. 261
. 225
. 50
. 62
. 243
I
ICC, see Internal Call Controller (ICC)
ICLID, see Caller ID
IGAR, see inter-gateway alternate routing (IGAR)
IGC, see Inter-Gateway Calls (IGC)
increased attendant consoles . . . . . . . . . . .
increased text field length for feature buttons - DCP
individual attendant access . . . . . . . . . . . .
individual operator access . . . . . . . . . . . .
integrated directory . . . . . . . . . . . . . . .
Integrated Services Digital Network (ISDN)
automatic Termination Endpoint Identifier (TEI) .
Basic Rate Interface (ISDN-BRI) . . . . . . . .
call-by-call service selection . . . . . . . . . .
ETSI functionality . . . . . . . . . . . . . . .
facility and non-facility associated signaling . . .
feature plus . . . . . . . . . . . . . . . . .
Multiple Subscriber Number (MSN) . . . . . .
NT interface on TN556C . . . . . . . . . . .
. 43
. 87
. 37
. 37
. 240
. 135
. 136
. 135
. 135
. 136
. 136
. 138
. 139
presentation restriction. . . . . . . . . . . . . 139
queue to best ISDN support . . . . . . . . . . . 65
trunks . . . . . . . . . . . . . . . . . . . . 135
wideband switching . . . . . . . . . . . . . . 139
integration with Cajun rules . . . . . . . . . . . . 155
intelligent networking . . . . . . . . . . . . . . . 147
intercom
automatic . . . . . . . . . . . . . . . . . . . . 80
automatic answer . . . . . . . . . . . . . . 80, 248
dial . . . . . . . . . . . . . . . . . . . . . . . 80
inter-gateway alternate routing (IGAR) . . . . . . . 123
Inter-Gateway Calls (IGC) . . . . . . . . . . . . . 162
internal automatic answer . . . . . . . . . . . . . 248
Internal Call Controller (ICC) . . . . . . . . . . 27, 192
international digital connectivity . . . . . . . . . . 135
International Standardization Organization (ISO) . . 164
Internet Protocol (IP) . . . . . . . . . . . 122, 129, 149
asynchronous links . . . . . . . . . . . . 176, 228
Avaya Softphone . . . . . . . . . . . . . . 83, 234
for pocket PC . . . . . . . . . . . . . . . . 84
H.323 trunk . . . . . . . . . . . . . . . . . . 129
improved button downloads . . . . . . . . . . 129
increased trunk capacity . . . . . . . . . . . . 129
loss groups . . . . . . . . . . . . . . . . . . 130
Port Network Connectivity (PNC) . . . . . . . . 123
report . . . . . . . . . . . . . . . . . . . . . 225
sending and receiving faxes . . . . . . . . . . 157
T.38 faxes over the Internet . . . . . . . . . . 159
trunks . . . . . . . . . . . . . . . . . . . . 130
fallback to PSTN . . . . . . . . . . . . . . 131
link bounce . . . . . . . . . . . . . . . . 131
TTY over Avaya IP trunks . . . . . . . . . . . 162
TTY pass through mode . . . . . . . . . . . . 163
TTY relay mode . . . . . . . . . . . . . . . . 163
inter-PBX attendant calls . . . . . . . . . . . . . . 39
Interruptible Aux work . . . . . . . . . . . . . . . . 52
intrusion . . . . . . . . . . . . . . . . . . . . . . 39
IP bearer duplication using the TN2602AP circuit pack190
bearer signal duplication . . . . . . . . . . . . 191
load balancing . . . . . . . . . . . . . . . . 190
reduced channels with duplication . . . . . . . 191
IP endpoint Time-to-Service (TTS) . . . . . . . . . 192
IP overload control . . . . . . . . . . . . . . . . 156
IP packet monitors . . . . . . . . . . . . . . . . 202
IP Softphone and IP Agent
RoadWarrior mode . . . . . . . . . . . . . 84, 233
Shared Control mode . . . . . . . . . . . . 84, 234
Telecommuter mode . . . . . . . . . . . . . 84, 234
ISDN, see Integrated Services Digital Network (ISDN)
ISDN-BRI, see Integrated Services Digital Network (ISDN),
Basic Rate Interface (ISDN-BRI)
ISO 8859-1 encoding compatibility . . . . . . . . . 262
ISO, see International Standardization Organization (ISO)
Italian Distributed Communications System (DCS) protocol
96, 148
Issue 6 May 2009
273
Index
J
LWC, see Leave Word Calling (LWC)
Japanese national private networking support . . 97, 139
Java Telephony Application Programming Interface (JTAPI)
34
JTAPI, see Java Telephony Application Programming
Interface (JTAPI)
M
K
katakana characters . . . . . . . . . . . . . . . . 97
L
LAI, see Look-Ahead Interflow (LAI)
last number dialed . . . . . . . . . . . . . . . . . 249
LDAP see Lightweight Directory Access Protocol (LDAP)
Least Occupied Agent (LOA) . . . . . . . . . . . . 65
Leave Word Calling (LWC) . . . . . . . . . . 107, 168
QSIG/DCS . . . . . . . . . . . . . . . . . . . 108
LEC, see Local Exchange Carrier (LEC)
license modes . . . . . . . . . . . . . . . . . . . 255
license-error mode . . . . . . . . . . . . . . . 255
license-normal mode . . . . . . . . . . . . . . 255
no-license mode . . . . . . . . . . . . . . . . 256
Lightweight Directory Access Protocol (LDAP) . . . . 217
Limit Number of Concurrent Calls (LNCC) . . . . . . 256
line load control . . . . . . . . . . . . . . . . . . 252
link recovery . . . . . . . . . . . . . . . . . . . . 124
Linux platforms
time of day clock synchronization . . . . . . . . 231
listed directory number . . . . . . . . . . . . . . . 41
Listen-only FAC for service observing . . . . . . . . 69
LNCC, see Limit Number of Concurrent Calls (LNCC)
LOA, see Least Occupied Agent (LOA)
local call timer automatic start/stop . . . . . . . . . 249
Local Exchange Carrier (LEC) . . . . . . 95, 143, 238
local exchange trunks . . . . . . . . . . . . . . . 144
800-service. . . . . . . . . . . . . . . . . . . 144
Central Office (CO) . . . . . . . . . . . . 133, 144
Digital Service 1 (DS1) . . . . . . . . . . . . . 144
Direct Inward Dialing (DID) . . . . . . . . . . . 144
Direct Inward/Outward Dialing (DIOD) . . . . . . 145
Foreign Exchange (FX) . . . . . . . . . . . . . 145
Wide Area Telecommunications Service (WATS) . 145
local feedback for queued ACD calls . . . . . . . . 53
local music-on-hold . . . . . . . . . . . . . . . . 229
Local Survivable Processor (LSP) . . . . . . . . 27, 192
localization . . . . . . . . . . . . . . . . . . . . 95
locally sourced announcements and music . . . . . 66
logged-in advocate agent counting . . . . . . . . . 55
long hold recall . . . . . . . . . . . . . . . . . . 249
Look-Ahead Interflow (LAI) . . . . . . . . . . . 57, 135
enhanced information forwarding . . . . . . . . 57
look-ahead routing . . . . . . . . . . . . . . 135, 184
loss groups for IP . . . . . . . . . . . . . . . . . 130
loudspeaker paging access . . . . . . . . . . . . . 81
LSP, see Local Survivable Processor (LSP)
274 Avaya AuraTM Communication Manager Overview
maintain external ring tone after internal transfer . . 245
making calls . . . . . . . . . . . . . . . . . . . . 44
malicious call trace . . . . . . . . . . . . . . 202, 228
logging . . . . . . . . . . . . . . . . . . 202, 228
over ETSI PRI. . . . . . . . . . . . . . . . . 202
management
calls . . . . . . . . . . . . . . . . . . . . . . 54
property . . . . . . . . . . . . . . . . . . . . 92
manual message waiting . . . . . . . . . . . . . 108
manual originating line service. . . . . . . . . . . 249
manual signaling . . . . . . . . . . . . . . . . . . 81
Manufacturer-Specific Information (MSI) . . . . . 65, 168
mask station name and number for internal calls . . 202
media encryption . . . . . . . . . . . . . . . . . 202
meet-me conferencing . . . . . . . . . . . . . . . 75
merge of IP Connect and Multiconnect configurations 153
message integration . . . . . . . . . . . . . . . 103
Message Sequence Tracer (MST) . . . . . . . . . 109
Message Waiting Indication (MWI) . . . . . . . . . 168
messages
audible message waiting . . . . . . . . . . . . 107
manual message waiting . . . . . . . . . . . . 108
retrieval . . . . . . . . . . . . . . . . . . . . 109
MFP, see Multi-Frequency Packet (MFP) signaling - Russia
misoperation handling . . . . . . . . . . . . . . 250
MLPP, see multiple level precedence and preemption
(MLPP)
MMCH, see Multimedia Call Handling (MMCH)
MMCX, see Multimedia Communications Exchange (MMCX)
mobility . . . . . . . . . . . . . . . . . . . . . . 113
modem over IP (MoIP) . . . . . . . . . . . . . . 158
MoIP, see modem over IP (MoIP)
monitoring calls. . . . . . . . . . . . . . . . . . . 44
MSI, see Manufacturer-Specific Information (MSI)
MSN, see Multiple Subscriber Number (MSN)
MST, see Message Sequence Tracer (MST)
multiappearance preselection and preference . . . 250
Multi-Frequency Packet (MFP) signaling - Russia . 97, 139
multi-location dial plans . . . . . . . . . . . . . . 171
multimedia, see multimedia calling
Multimedia Call Handling (MMCH) . . . . . . . . 78, 178
multimedia calling
Application Server Interface (ASI) . . . . . . 78, 177
call early answer on vectors and stations . . . 78, 177
call redirection to multimedia endpoint . . . . 79, 179
data conferencing . . . . . . . . . . . . . . . . 77
(T.120) via ESM . . . . . . . . . . . . . 79, 179
Expansion Services Module (ESM) . 77, 79, 177, 179
hold, conference, transfer, and drop . . . . . 79, 179
multiple-port networks . . . . . . . . . . . . . 179
queuing with voice announcement . . . . . . . . 79
voice and video . . . . . . . . . . . . . . . . . 77
Multimedia Communications Exchange (MMCX) . . . 78
Index
Multinational Locations . . . . . . . . . . . . . . . 98
analog line board parameters per location . . . . 99
companding for DCP telephones and circuit packs per
location . . . . . . . . . . . . . . . . . . . . 99
location ID in Call Detail Record (CDR) records . . 99
loss plans per location . . . . . . . . . . . . . 100
multifrequency signaling per trunk group . . . . . 100
multiple call handling (forced) . . . . . . . . . . . . 65
multiple level precedence and preemption (MLPP) . . 251
announcements for precedence calling . . . . . . 251
dual homing . . . . . . . . . . . . . . . . . . 252
end office access line hunting . . . . . . . . . . 252
line load control . . . . . . . . . . . . . . . . . 252
precedence call waiting . . . . . . . . . . . . . 252
precedence calling . . . . . . . . . . . . . . . 252
precedence routing . . . . . . . . . . . . . . . 253
preemption . . . . . . . . . . . . . . . . . . . 253
worldwide numbering and dialing plan (WNDP) . . 254
multiple location support . . . . . . . . . . . . . . 184
for network regions . . . . . . . . . . . . 151, 185
multiple music sources . . . . . . . . . . . . . . . 229
multiple music/audio sources . . . . . . . . . . . . 66
multiple network regions per CLAN . . . . . . 151, 193
multiple split queuing . . . . . . . . . . . . . . . 66, 67
Multiple Subscriber Number (MSN) . . . . . . . . . 138
Multi-Tech gateway support . . . . . . . . . . . . 29
music-on-hold . . . . . . . . . . . . . . . . . . . 228
MWI, see Message Waiting Indication (MWI)
N
name and number identification . . . . . . . . . . . 168
name display on unsupervised transfer . . . . . . . 166
name/number permanent display . . . . . . . . . . 118
names registration . . . . . . . . . . . . . . . . . 91
NAPT, see Network Address Port Translation (NAPT)
NAT, see Network Address Translation (NAT)
national private networking support - Japan . . . 97, 139
NCR, see Network Call Redirection (NCR)
Network Address Port Translation (NAPT) . . . . . . 161
Network Address Translation (NAT) . . . . . . . . . 160
with shuffling . . . . . . . . . . . . . . . . . . 160
network answer supervision . . . . . . . . . . . . 185
Network Call Redirection (NCR) . . . . . . . . . 47, 66
2B-channel transfer . . . . . . . . . . . . . . . 67
Network Region Wizard (NRW) . . . . . . . . . . . 123
network regions . . . . . . . . . . . . . . . . . . 151
multiple location support for . . . . . . . . 151, 185
network services . . . . . . . . . . . . . . . . . . 135
night service . . . . . . . . . . . . . . . . . . . . 254
enhanced . . . . . . . . . . . . . . . . . . . 254
no dial tone conferencing . . . . . . . . . . . . . . 75
no hold conference. . . . . . . . . . . . . . . . . 75
node number routing . . . . . . . . . . . . . . . . 184
NRW, see Network Region Wizard (NRW)
NT interface on TN556C . . . . . . . . . . . . . . 139
O
Octel integration . . . . . . . . . . . .
off-PBX station (OPS). . . . . . . . . .
off-premises station . . . . . . . . . . .
on-hook programming . . . . . . . . .
Open System Interconnect (OSI) . . . .
operator dial access . . . . . . . . . .
OPS, see off-PBX station (OPS)
optional software . . . . . . . . . . . .
Options to clear display of collected digits
OSI, see Open System Interconnect (OSI)
Overall Loudness Rating . . . . . . . .
override of diversion features . . . . . .
overview of Communication Manager . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
109, 167
. . . 115
. . 234
. . 235
. . 135
. . . 37
. . . . . . 26
. . . . . . 62
. . . . . 130
. . . . . . 41
. . . . . . 25
P
packet error history report . . . . . . . . . . . . . 226
paging access
loudspeaker . . . . . . . . . . . . . . . . . . . 81
parsing capabilities of the history report . . . . . . 227
pass advice of charge information to world class BRI
endpoints . . . . . . . . . . . . . . . . . . . . 179
pass advice of charge to BRI endpoints . . . . . . 218
PASTE, see PC Application Software Translation Exchange
(PASTE)
PAT, see Port Address Translation (PAT)
path replacement with path retention . . . . . . . . 169
PC Application Software Translation Exchange (PASTE)67
PCOL, see Personal Central Office Line (PCOL)
PE, see Processor Ethernet (PE)
pending work mode change . . . . . . . . . . . . . 49
per call CPN restriction . . . . . . . . . . . . . . 207
per line CPN restriction . . . . . . . . . . . . . . 208
percent allocation distribution . . . . . . . . . . . . 56
Percentage allocation routing . . . . . . . . . . . . 60
Periodic Pulse Metering (PPM) . . . . . . . . . . 218
Permanent Virtual Paths (PVP) . . . . . . . . . . 121
Personal Central Office Line (PCOL) . . . . . . . . 140
Personal Station Access (PSA) . . . . . . . . . . . 117
name/number permanent display . . . . . . . . . 118
personalized ringing . . . . . . . . . . . . . . . 257
placing calls . . . . . . . . . . . . . . . . . . . . 44
PMS, see Property Management System (PMS)
PNA, see Private Network Access (PNA)
PNC, see port network connectivity (PNC)
Port Address Translation (PAT) . . . . . . . . . . 161
port network and gateway connectivity . . . . . . . 121
port network and link usage report . . . . . . . . . 226
Port Network Connectivity (PNC)
Asynchronous Transfer Mode (ATM) . . . . . . 121
over WAN . . . . . . . . . . . . . . . . . 121
Internet Protocol (IP) . . . . . . . . . . . . . 123
posted messages . . . . . . . . . . . . . . . . . 257
power failure transfer . . . . . . . . . . . . . . . 193
PPM, see Periodic Pulse Metering (PPM)
Issue 6 May 2009
275
Index
PPN, see Processor Port Network (PPN)
precedence call waiting . . . . . . . . . . . . . . 252
precedence calling . . . . . . . . . . . . . . . . . 252
precedence routing . . . . . . . . . . . . . . . . 253
preemption . . . . . . . . . . . . . . . . . . . . 253
presentation restriction . . . . . . . . . . . . . . . 139
PRI, see Primary Rate Interface (PRI)
Primary Rate Interface (PRI) . . . . . . . . . 127, 135
priority calling . . . . . . . . . . . . . . . . . . . 257
priority queue . . . . . . . . . . . . . . . . . 42, 167
privacy
attendant lockout . . . . . . . . . . . . . . 39, 198
auto exclusion . . . . . . . . . . . . . . . . . 210
manual exclusion . . . . . . . . . . . . . . . . 210
Private Network Access (PNA) . . . . . . . . . . . 148
Processor Ethernet (PE) . . . . . . . . . . . . . . 152
adjuncts . . . . . . . . . . . . . . . . . . . . 152
H.248 and H.323 registration . . . . . . . . . . 153
S8500 Media Servers . . . . . . . . . . . . . . 154
processor occupancy report . . . . . . . . . . . . 226
Processor Port Network (PPN) . . . . . . . . . . . 121
Property Management System (PMS) . . . . . . 90, 176
digit to insert/delete . . . . . . . . . . . . . . . 92
interface . . . . . . . . . . . . . . . . . . . . 92
PSA, see Personal Station Access (PSA)
PSTN, see Public Switched Telephone Network (PSTN)
public network call priority . . . . . . . . . . . . . 100
public networking and connectivity . . . . . . . . . 143
Public Switched Telephone Network (PSTN) . . . . . 66
pull transfer . . . . . . . . . . . . . . . . . . . . 258
PVP, see Permanent Virtual Paths (PVP)
Q
QoS, see Quality of Service (QoS)
QSIG . . . . . . . . . . . . . . . . . . . . . .
basic . . . . . . . . . . . . . . . . . . . . .
call completion . . . . . . . . . . . . . . . .
call forwarding (diversion) . . . . . . . . . . .
Call Independent Signaling Connections (CISC)
call offer . . . . . . . . . . . . . . . . . . .
call transfer. . . . . . . . . . . . . . . . . .
called name ID . . . . . . . . . . . . . . . .
Centralized Attendant Service (CAS) . . . . . .
attendant return call . . . . . . . . . . . .
priority queue . . . . . . . . . . . . . . .
RLT emulation through a PRI . . . . . . . .
Class of Restriction (COR), attendant display . .
Communication Manager/Octel integration . . .
Leave Word Calling (LWC) . . . . . . . . . .
Manufacturer-Specific Information (MSI) . . . .
Message Waiting Indication (MWI) . . . . . . .
name and number identification . . . . . . . .
overview . . . . . . . . . . . . . . . . . . .
path replacement with path retention . . . . . .
reroute after diversion to voice mail . . . . . .
. 135
. 164
. 164
. 165
. 166
. 166
. 166
. 166
. 167
. 167
. 167
. 167
. 167
. 167
. 108
. 168
. 168
. 168
. 164
. 169
. 169
276 Avaya AuraTM Communication Manager Overview
stand-alone path replacement . . . . . . . . . 169
supplementary services and rerouting . . . . . 170
support for Unicode . . . . . . . . . . . . . . . 88
VALU . . . . . . . . . . . . . . . . . . . . . 170
call coverage . . . . . . . . . . . . . . . 170
call coverage and CAS . . . . . . . . . . . 170
distinctive alerting . . . . . . . . . . . . . 170
QSIG Supplementary Service - Advice of Charge (SS-AOC)
145
QSIG/DCS
Leave Word Calling (LWC) . . . . . . . . . . . 108
voice mail interworking. . . . . . . . . 110, 148, 169
Quality of Service (QoS) . . . . . . . . . . . 149, 154
802.1p/Q . . . . . . . . . . . . . . . . . . . 154
call control . . . . . . . . . . . . . . . . . . 156
codecs . . . . . . . . . . . . . . . . . . . . 155
differentiated services (DiffServ) . . . . . . . . 155
dynamic jitter buffers . . . . . . . . . . . . . 155
for VoIP . . . . . . . . . . . . . . . . . . . . 156
integration with Cajun rules . . . . . . . . . . 155
RSVP . . . . . . . . . . . . . . . . . . . . 157
shuffling and hairpinning . . . . . . . . . . . . 160
to endpoints . . . . . . . . . . . . . . . . . . 157
variable length ping . . . . . . . . . . . . . . 163
Variable Length Subnet Mask (VLSM) . . . . . 164
queue status indications . . . . . . . . . . . . . . 54
queue to best ISDN support . . . . . . . . . . . . . 65
queuing
multiple split. . . . . . . . . . . . . . . . . . . 66
with voice announcement . . . . . . . . . . . . 79
R
real-time reports . . . . . . . . . . . . . . . . . . 54
reason codes . . . . . . . . . . . . . . . . . . . . 68
recall . . . . . . . . . . . . . . . . . . . . . . . 37
recall signaling . . . . . . . . . . . . . . . . . . 258
recalling the attendant . . . . . . . . . . . . . . . 37
recent change history report . . . . . . . . . . . . 226
record on messaging . . . . . . . . . . . . . . . 106
recorded telephone dictation access . . . . . . . . 258
redirection of calls . . . . . . . . . . . . . . . . 240
call forward busy/do not answer . . . . . . . . 240
call forwarding all calls . . . . . . . . . . . . . 241
call forwarding override . . . . . . . . . . . . 241
call redirection intervals . . . . . . . . . . . . 242
on no answer . . . . . . . . . . . . . . . . . . 68
RedSky Technologies, see E911 device location for IP
telephones
refresh route report . . . . . . . . . . . . . . . . 226
Release Link Trunks (RLT) . . . . . . . . . . 140, 167
emulation through a PRI . . . . . . . . . . . . 167
release loop operation . . . . . . . . . . . . . . . 42
reliability and survivability . . . . . . . . . . . . . 187
remote access trunks . . . . . . . . . . . . . 140, 234
remote logout of agent . . . . . . . . . . . . . . . 68
reports . . . . . . . . . . . . . . . . . . . . . . 225
Index
attendant position . . . . . . . . . . . . . . . . 225
blockage study . . . . . . . . . . . . . . . . . 225
call coverage . . . . . . . . . . . . . . . . . . 225
coverage points . . . . . . . . . . . . . . . . 225
display ARP . . . . . . . . . . . . . . . . . . 225
emergency and journal . . . . . . . . . . . . . 225
historical . . . . . . . . . . . . . . . . . . . . 54
parsing capabilities of . . . . . . . . . . . . 227
hunt group measurements. . . . . . . . . . . . 225
Internet Protocol (IP) . . . . . . . . . . . . . . 225
management . . . . . . . . . . . . . . . . . . 54
packet error history . . . . . . . . . . . . . . . 226
port network and link usage . . . . . . . . . . . 226
processor occupancy . . . . . . . . . . . . . . 226
real-time . . . . . . . . . . . . . . . . . . . . 54
recent change history . . . . . . . . . . . . . . 226
refresh route . . . . . . . . . . . . . . . . . . 226
summary . . . . . . . . . . . . . . . . . . . . 226
tandem traffic . . . . . . . . . . . . . . . . . . 226
traffic . . . . . . . . . . . . . . . . . . . . . 226
trunk group detailed measurement . . . . . . . . 227
reroute after diversion to voice mail . . . . . . . . . 169
reserve agent time in queue activation . . . . . . . 56
reset shift call . . . . . . . . . . . . . . . . . . . 258
Resource Reservation Protocol (RSVP) . . . . . . . 157
restriction - controlled . . . . . . . . . . 204, 210, 229
ringback queuing . . . . . . . . . . . . . . . . . 259
ringer cutoff . . . . . . . . . . . . . . . . . . . . 259
ringing
abbreviated and delayed . . . . . . . . . . . . 259
distinctive . . . . . . . . . . . . . . . . . . . 245
options . . . . . . . . . . . . . . . . . . . . . 259
personalized . . . . . . . . . . . . . . . . . . 257
RLT, see Release Link Trunks (RLT)
room numbers dialing plan . . . . . . . . . . . . . 92
room status . . . . . . . . . . . . . . . . . . . 38, 89
routing features . . . . . . . . . . . . . . . . . . 181
RSVP, see Resource Reservation Protocol (RSVP)
Russian CO support on G700 Media Gateway . . 97, 133
Russian Multi-Frequency Packet (MFP) signaling 97, 139
S
satellite hops . . . . . . . . . . . . . . . . . . . 148
SBS, see Separation of Bearer and Signaling (SBS)
scheduling . . . . . . . . . . . . . . . . . . . . . 230
secure shell and secure FTP (SSH/SFTP) . . . . . . 205
security of IP telephone config files . . . . . . . . . 205
security of IP telephone registration/H.323 signaling channel
205
Security Violation Notification (SVN) . . . . . 206, 230
security, privacy, and safety . . . . . . . . . . . . 197
end user . . . . . . . . . . . . . . . . . . . . 207
backup alerting . . . . . . . . . . . . . . . 207
barrier codes . . . . . . . . . . . . . . . . 207
crisis alert to a digital numeric pager . . . . . 208
crisis alert to a digital station . . . . . . . . . 208
crisis alert to an attendant console . . . . . 209
emergency access to the attendant . . . . . 209
per call CPN restriction . . . . . . . . . . . 207
per line CPN restriction . . . . . . . . . . . 208
privacy, auto exclusion . . . . . . . . . . . 210
privacy, manual exclusion. . . . . . . . . . 210
restriction - controlled . . . . . . . . . . . . 210
station lock . . . . . . . . . . . . . . . . 210
station lock by Time of Day . . . . . . . . . . 211
system administrator . . . . . . . . . . . . . 197
access security gateway (ASG) . . . . . . . 197
alternate facility restriction levels . . . . . . 198
alternate operations support system alarm . . 198
call restrictions . . . . . . . . . . . . . . . 199
Class of Restriction (COR) . . . . . . . . . 199
customer-provided equipment alarm . . . . . 199
data privacy . . . . . . . . . . . . . . . . 199
data restriction . . . . . . . . . . . . . . . 200
encryption algorithm for bearer channels . . . 200
facility restriction levels and traveling class marks201
H.248 link encryption . . . . . . . . . . . . 201
malicious call trace . . . . . . . . . . . . . 202
mask station name and number for internal calls202
media encryption . . . . . . . . . . . . . . 202
privacy - attendant lockout . . . . . . . . . 198
restriction - controlled . . . . . . . . . . . . 204
Security Violation Notification (SVN) . . . . . 206
signaling encryption for SIP trunks . . . . . 206
SRTP media encryption . . . . . . . . . . 200
station security codes . . . . . . . . . . . 206
tripwire security . . . . . . . . . . . . . . 206
select line appearance conferencing . . . . . . . . . 76
selective conference mute . . . . . . . . . . . 42, 77
selective conference party display and drop . . . . . 76
self-administered telephones . . . . . . . . . . . 263
send all calls . . . . . . . . . . . . . . . . . . . 259
separate licensing for TDM telepones and TDM trunks128
Separation of Bearer and Signaling (SBS) . . . . . 125
serial calling . . . . . . . . . . . . . . . . . . . . 42
service observing . . . . . . . . . . . . . . . . . . 68
by COR . . . . . . . . . . . . . . . . . . . . . 69
of VDNs . . . . . . . . . . . . . . . . . . . . 69
remote . . . . . . . . . . . . . . . . . . . . . 69
vector-initiated . . . . . . . . . . . . . . . . . 70
Session Iniatiation Protocol (SIP)
SIP Visiting User . . . . . . . . . . . . . . . . 118
Session Initiation Protocol (SIP) . . . . . . 116, 131, 155
trunks . . . . . . . . . . . . . . . . . . . . 132
shuffling . . . . . . . . . . . . . . . . . . . . . 160
and NAT devices . . . . . . . . . . . . . . . 160
signaling encryption for SIP trunks . . . . . . . . . 206
single-digit dialing and mixed station numbering . . . 92
SIP, see Session Initiation Protocol (SIP)
site statistics for remote port networks . . . . . . . . 70
six party conferencing. . . . . . . . . . . . . . . . 73
Issue 6 May 2009
277
Index
skill . . . . . . . . . . . . . . . . . . . . . . . . 64
SLS, see Standard Local Survivability (SLS)
SMS, see system management service (SMS)
sniffers . . . . . . . . . . . . . . . . . . . . . . 202
special dial tone . . . . . . . . . . . . . . . . . . 260
SREPN, see Survivable Remote Expansion Port Network
(SREPN)
SRTP media encryption . . . . . . . . . . . . . . 200
SS-AOC, see QSIG Supplementary Service - Advice of
Charge (SS-AOC)
SSH/SFTP, see secure shell and secure FTP (SSH/SFTP)
stand-alone path replacement . . . . . . . . . . . 169
Standard Local Survivability (SLS) . . . . . . . . . 194
station hunt before coverage . . . . . . . . . . . . 260
station hunting . . . . . . . . . . . . . . . . . . . 260
station lock . . . . . . . . . . . . . . . . . . . . 210
station lock by Time of Day . . . . . . . . . . . . . 211
station security codes . . . . . . . . . . . . 206, 230
station self display . . . . . . . . . . . . . . . . . 260
station used as a virtual extension . . . . . . . . . 261
suite check-in . . . . . . . . . . . . . . . . . . . 93
summary report . . . . . . . . . . . . . . . . . . 226
supplementary services and rerouting . . . . . . . . 170
supplementary services, definition of . . . . . . . . 164
support for the HP DL380G2 server . . . . . . . . . 261
Survivable Remote Expansion Port Network (SREPN) 194
SVC, see Switched Virtual Circuits (SVC)
SVN, see Security Violation Notification (SVN)
switch
Asynchronous Transfer Mode (ATM) . . . . 121, 127
CSS (direct connect) . . . . . . . . . . . . . . 121
Switched Virtual Circuits (SVC) . . . . . . . . . . . 121
system management . . . . . . . . . . . . . . . . 213
system management service (SMS) . . . . . . . . . 32
T
T.120 protocols . . . . . . . . . . . . . . . . 79, 179
T.38 faxes over the Internet. . . . . . . . . . . . . 159
T1 digital interface . . . . . . . . . . . . . . 128, 144
tandem switch . . . . . . . . . . . . . . . . . . . 149
tandem through . . . . . . . . . . . . . . . . . . 149
Tandem Tie-Trunk Network (TTTN) . . . . . . . . . 149
tandem traffic report . . . . . . . . . . . . . . . . 226
TDD, see Telecommunication Devices for the Deaf (TDD);
see also TTY
team button . . . . . . . . . . . . . . . . . . . . 261
TEI, see Termination Endpoint Identifier (TEI)
Telecommunication Devices for the Deaf (TDD) . . . 161
telecommuting access . . . . . . . . . . . . . . . 233
telecommuting and remote office . . . . . . . . . . 233
telephone display . . . . . . . . . . . . . . . . . 262
ISO 8859-1 encoding compatibility . . . . . . . . 262
telephones
announcements . . . . . . . . . . . . . . . . 214
digital
2420 DCP . . . . . . . . . . . . . . . . . . 110
278 Avaya AuraTM Communication Manager Overview
self-administration . . . . . . . . . . . . . . . 263
telephony . . . . . . . . . . . . . . . . . . . . 235
telephony service (TS) . . . . . . . . . . . . . . . 32
Telephony Services Application Programming Interface
(TSAPI) . . . . . . . . . . . . . . . . . . . . . 34
temporary bridged appearance . . . . . . . . . . 263
Temporary Signaling Connection (TSC) . . . . . . 164
tenant partitioning . . . . . . . . . . . . . . 229, 230
Terminal Translation Initialization (TTI) . . . . . 119, 231
terminating extension group . . . . . . . . . . . . 263
Termination Endpoint Identifier (TEI), automatic . . 135
three party conferencing . . . . . . . . . . . . . . 73
tie trunks . . . . . . . . . . . . . . . . . . . . . 140
time of day . . . . . . . . . . . . . . . . . . . . 240
time of day clock synchronization
Linux platforms . . . . . . . . . . . . . . . . 231
UNIX platforms . . . . . . . . . . . . . . . . 231
via LAN source . . . . . . . . . . . . . . . . 231
time of day routing . . . . . . . . . . . . . . 184, 264
timed automatic disconnect for outgoing trunk calls . 140
timed call disconnection for outgoing trunk calls . . 264
timed reminders . . . . . . . . . . . . . . . . . . 42
TN464GP/TN2464BP universal DS-1 circuit pack . . 128
TN556C circuit pack . . . . . . . . . . . . . . . 139
TN787 . . . . . . . . . . . . . . . . . . . . . 79, 179
TOS, see Type-Of-Service (TOS)
traffic report . . . . . . . . . . . . . . . . . . . 226
transfer . . . . . . . . . . . . . . . . . . . . . 264
abort . . . . . . . . . . . . . . . . . . . . . 265
outgoing trunk to outgoing trunk . . . . . . . . 265
recall . . . . . . . . . . . . . . . . . . . . . 265
trunk-to-trunk . . . . . . . . . . . . . . . . . 265
upon hang-up . . . . . . . . . . . . . . . . . 265
TransTalk 9000 digital wireless system. . . . . . . . 119
traveling class marks . . . . . . . . . . . . . . . 185
tripwire security. . . . . . . . . . . . . . . . . . 206
trunk call disconnection . . . . . . . . . . . . 140, 264
trunk connectivity . . . . . . . . . . . . . . . . . 127
trunk flash . . . . . . . . . . . . . . . . . . . . 266
trunk group
attendant control of access . . . . . . . . . . . . 44
busy/warning indicators to attendant . . . . . . . 46
circuits . . . . . . . . . . . . . . . . . . . . 231
detailed measurement report . . . . . . . . . . 227
identification . . . . . . . . . . . . . . . . . . 49
trunk identification by attendant . . . . . . . . . . . 46
trunk signaling and error recovery . . . . . . . . . 149
trunks
analog
Caller ID (ICLID) . . . . . . . . . . . . 143, 243
auxiliary . . . . . . . . . . . . . . . . . . . 132
Advanced Private Line Termination (APLT) . 132
echo cancellation. . . . . . . . . . . . 128, 143
digital . . . . . . . . . . . . . . . . . . . . . 133
Caller ID (ICLID) . . . . . . . . . . . . 143, 243
Digital Service 1 (DS1). . . . . . . . . . . 127, 143
Index
Direct Inward Dialing (DID) . . . . . . . . . . . 134
Direct Inward/Outward Dialing (DIOD) . . . . . . 134
Foreign Exchange (FX) . . . . . . . . . . . . . 134
group circuits . . . . . . . . . . . . . . . . . . 231
H.323 . . . . . . . . . . . . . . . . . . . . . 129
Internet Protocol (IP) . . . . . . . . . . . . . . 130
ISDN. . . . . . . . . . . . . . . . . . . . . . 135
local exchange . . . . . . . . . . . . . . . . . 144
800-service . . . . . . . . . . . . . . . . . 144
Central Office (CO) . . . . . . . . . . 133, 144
Digital Service 1 (DS1) . . . . . . . . . . . . 144
Direct Inward Dialing (DID) . . . . . . . . . . 144
Direct Inward/Outward Dialing (DIOD) . . . . 145
Foreign Exchange (FX) . . . . . . . . . . . 145
Wide Area Telecommunications Service (WATS)145
Personal Central Office Line (PCOL) . . . . . . . 140
Release Link (RLT) . . . . . . . . . . . . 140, 167
remote access . . . . . . . . . . . . . . 140, 234
tandem. . . . . . . . . . . . . . . . . . . . . 148
tie . . . . . . . . . . . . . . . . . . . . . . . 140
Wide Area Telecommunications Service (WATS) . 141
trunk-to-trunk transfer . . . . . . . . . . . . . . . 265
TS, see telephony service (TS)
TSAPI, see Telephony Services Application Programming
Interface (TSAPI)
TSC, see Temporary Signaling Connection (TSC)
TTI, see Terminal Translation Initialization (TTI)
TTS, see IP endpoint Time-to-Service (TTS)
TTTN, see Tandem Tie-Trunk Network (TTTN)
TTY . . . . . . . . . . . . . . . . . . . . . . . . 161
over analog and digital trunks . . . . . . . . . . 162
over Avaya IP trunks . . . . . . . . . . . . . . 162
pass through mode . . . . . . . . . . . . . . . 163
relay mode . . . . . . . . . . . . . . . . . . . 163
Type-Of-Service (TOS) . . . . . . . . . . . . . . . 155
U
UCD, see Uniform Call Distribution (UCD)
UDP, see Uniform Dial Plan (UDP)
UDS1 circuit pack with echo cancellation .
Unicode support . . . . . . . . . . . . .
QSIG . . . . . . . . . . . . . . . .
Uniform Call Distribution (UCD) . . . . . .
Uniform Dial Plan (UDP) . . . . . . . . .
UNIX platforms
time of day clock synchronization . . .
usage allocation . . . . . . . . . . . . .
user service . . . . . . . . . . . . . . .
User-to-User Information (UUI)
over the public network . . . . . . . .
propagation . . . . . . . . . . . . .
UUI, see User-to-User Information (UUI)
VCO, see Voice Carry Over (VCO)
VDN, see Vector Directory Number (VDN)
vector commands. . . . . . . . . . . . . . . . . . 59
Vector Directory Number (VDN) . . . . . . . 60, 64, 68
display VDN for route-to DAC . . . . . . . . . . 61
in a coverage path . . . . . . . . . . . . . . . . 61
observing on agent answer . . . . . . . . . . . 70
of origin announcement . . . . . . . . . . . . . 61
override for ASAI messages . . . . . . . . . . . 50
return destination . . . . . . . . . . . . . . . . 61
vectoring . . . . . . . . . . . . . . . . . . . . . . 59
attendant . . . . . . . . . . . . . . . . . . . . 40
holiday . . . . . . . . . . . . . . . . . . . . . 60
vector-initiated service observing . . . . . . . . . . 70
VIAS, see Visually Impaired Attendant Service (VIAS)
video . . . . . . . . . . . . . . . . . . . . . . 177
VIP wakeup . . . . . . . . . . . . . . . . . . . . 93
Virtual LAN (VLAN) . . . . . . . . . . . . . . . . 154
Visually Impaired Attendant Service (VIAS). . . . . . 46
VLSM, see Variable Length Subnet Mask (VLSM)
VMON, see VoIP Monitoring Manager (VMON)
Voice Announcement over LAN (VAL) Manager . . 216
Voice Carry Over (VCO) . . . . . . . . . . . . . 161
voice mail integration . . . . . . . . . . . . . . . . 78
voice mail interworking . . . . . . . . . . . . . . 169
QSIG/DCS . . . . . . . . . . . . . . . . 110, 148
voice mail retrieval button . . . . . . . . . . . . . . 110
voice mail system (VMS) . . . . . . . . . . . . . 254
voice message retrieval . . . . . . . . . . . . . . . 110
voice messaging and call coverage . . . . . . . . . 111
Voice Response Integration (VRI) . . . . . . . . . . 71
VoIP Monitoring Manager (VMON) . . . . . . . 147, 217
VRI, see Voice Response Integration (VRI)
VuStats . . . . . . . . . . . . . . . . . . . . . . 71
login IDs . . . . . . . . . . . . . . . . . . . . 71
service level. . . . . . . . . . . . . . . . . . . 71
W
.
.
.
.
.
.
.
.
.
.
128, 143
. . . 87
. . . 88
. . . 239
. . . 171
. . . . . 231
. . . . . 135
. . . . . 33
. . . . . 70
. . . . . 49
V
variable length ping . . . . . . . . . . . . . 163, 232
Variable Length Subnet Mask (VLSM) . . . . . 164, 232
wakeup
activation via confirmation tones . . . . . . . . . 93
automatic . . . . . . . . . . . . . . . . . . . . 89
daily . . . . . . . . . . . . . . . . . . . . . . 90
dual . . . . . . . . . . . . . . . . . . . . . . 91
VIP . . . . . . . . . . . . . . . . . . . . . . . 93
WATS, see Wide Area Telecommunications Service
(WATS)
Web services . . . . . . . . . . . . . . . . . . . . 32
whisper page . . . . . . . . . . . . . . . . . . . . 81
Wide Area Telecommunications Service (WATS)141, 145,
182
wideband switching . . . . . . . . . . . . . . . . 139
wireless
X-station mobility . . . . . . . . . . . . . . . . 119
WNDP, see worldwide numbering and dialing plan (WNDP)
world class tone detection . . . . . . . . . . . . . 101
worldwide numbering and dialing plan (WNDP) . . . 254
Issue 6 May 2009
279
Index
X
Xiox call accounting . . . . . . . . . . . . . . . . 93
XOIP Tone Detection Bypass . . . . . . . . . . . . 101
X-station mobility . . . . . . . . . . . . . . . . . 119
280 Avaya AuraTM Communication Manager Overview
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertising