advertisement
ACM Administration Manual
MAX Communication Server
Release 8.0
January 26, 2016
WARNING! Toll fraud is committed when individuals unlawfully gain access to customer telecommunication systems. This is a criminal offense. Currently, we do not know of any telecommunications system that is immune to this type of criminal activity. AltiGen
®
Communications,
Inc. will not accept liability for any damages, including long distance charges, which result from unauthorized and/or unlawful use. Although AltiGen Communications, Inc. has designed security features into its products, it is your sole responsibility to use the security features and to establish security practices within your company, including training, security awareness, and call auditing.
NOTICE: While every effort has been made to ensure accuracy, AltiGen Communications, Inc., will not be liable for technical or editorial errors or omissions contained within the documentation. The information contained in this documentation is subject to change without notice.
This documentation may be used only in accordance with the terms of the AltiGen Communications, Inc.,
License Agreement.
MAX Communication Server, MaxAdministrator, MaxCommunicator, MaxAgent, MaxSupervisor,
MaxOutlook, MaxInSight, MaxCall, AltiServ
®
, AltiLink, AltiConsole, VRPlayer, Zoomerang, IPTalk, Alti-
Mobile Extension, InTouch Dialer, and AltiReport are trademarks or registered trademarks of AltiGen
Communications, Inc. All other brand names mentioned are trademarks or registered trademarks of their respective manufacturers.
AltiGen's products are protected under one or more of the following U.S. patents, with other U.S. patents pending: 6532230; 6192344; 6292549; 6493439; 6909780; 6738465; 6754202; 6766006; 6928078;
6909709; 6956848; 7058047; 7013007; 7027578; 7280649; 7308092.
AltiGen Communications, Inc.
679 River Oaks Parkway
San Jose, CA 95134
Telephone: 888-AltiGen (258-4436)
Fax: 408-597-9020
E-mail: [email protected]
Web site: www.altigen.com
Copyright © AltiGen Communications, Inc. 2016. All rights reserved.
Contents
A
T
D
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
C
1
New Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3
Enhancements Included in Release 8.0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Enhancements Included in Release 7.5 Patch 1 . . . . . . . . . . . . . . . . . . . . . . 5
Enhancements Included in Release 7.5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Features No Longer Supported in Release 7.5 . . . . . . . . . . . . . . . . . . . . . . 8
Additional Changes in Release 7.5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Enhancements Included in Release 7.0 Update 1 . . . . . . . . . . . . . . . . . . . . . 9
Enhancements Included in Release 7.0 . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Enhancements Included in Release 6.7 Update 1 . . . . . . . . . . . . . . . . . . . . 11
Enhancements Included in Release 6.7 . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Enhancements Included in Release 6.5 Update 1 . . . . . . . . . . . . . . . . . . . . 12
Enhancements Included in Release 6.5 . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
Enhancements Included in Release 6.0 . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
C
2
System Requirements and Installation . . . . . . . . . . . . . . . . . . . . .29
Minimum System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Virtual Server Requirements: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
MAXCS Client Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Boards Supported in MAXCS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Installing MAX Communication Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Softswitch System Installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Installing MAXCS Administrator on a Network Client . . . . . . . . . . . . . . . . . . 38
C
3
Getting Around MaxAdministrator . . . . . . . . . . . . . . . . . . . . . . . . .39
Changing the Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
The Main Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Quick Access Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Boards View Window. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
Extension View Window. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
Trunk View Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Call Log View Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
MAXCS Administration Manual i
Workgroup View Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
Current Resource Statistics Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
Assigning Seat-Based Client Licenses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Stopping the AltiGen Switching Service . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Programs Available from the Windows Start Menu . . . . . . . . . . . . . . . . . . . . 48
C
4
System Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Setting General Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Setting a System Number Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
Updating the Dates of Annual Holidays . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Configuring System Speed Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Adding Speed Dial Entries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
Editing Speed Dial Entries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
Defining System Call Restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
Blocking Calls to Area Codes from All Extensions . . . . . . . . . . . . . . . . . . . 61
Setting Unrestricted Area Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Locking Attacked Extensions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Blocking All Outgoing Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Enabling Hop Off for Tie Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Setting 10-Digit Dialing Area Codes. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Adding and Deleting Account Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Internal Database Configuration (Internal Log Service) . . . . . . . . . . . . . . . 64
External (Remote) Logging of Call Data . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Exporting Through a Local Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Country-Relevant Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Setting Emergency Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Dialing Plan Rules for Non-North American Country. . . . . . . . . . . . . . . . . . 66
Audio Peripheral Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Configuring Music On Hold and Recorded Announcements . . . . . . . . . . . . . 68
RTP Resource Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Setting Greeting and Update Prompts . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Configuring Overhead Paging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Adding Feature Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
C
5
Voice Mail Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Setting Message Notification Retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
ii MAXCS Administration Manual
Setting Message Management Options . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Setting Message Recording Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Setting Exchange Integration Options . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Setting E-mail Messaging Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Defining a Distribution List. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
C
6
Auto Attendant Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . .79
Example: AA Planning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
Configuring Auto Attendants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Configuring Menu Items . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
Making Auto Attendant Assignments . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Phrase Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Using Pre-Recorded Prompts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Recording Custom Phrases from the AltiGen Phone. . . . . . . . . . . . . . . . . . 86
Using Professionally Recorded Phrases . . . . . . . . . . . . . . . . . . . . . . . . . . 86
C
7
Multilingual Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .89
Creating Language Phrase Packages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
Storing Language Phrase Packages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
Configuring for a Multilingual System . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
Enabling Multilingual Support in the Auto Attendant . . . . . . . . . . . . . . . . . . 91
Configuring the Extension . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Extension User Can Change Language Setting . . . . . . . . . . . . . . . . . . . . . 93
Configuring Feature Code #12 for Language Selection . . . . . . . . . . . . . . . 93
Using DNIS to Set the Language . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
Which Language Will Be Used? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
C
8
Call Recording Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . .97
Description of the Recorded File Name . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
Using a Remote Shared Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
C
9
Application Extension Configuration . . . . . . . . . . . . . . . . . . . . . . .103
Readying the Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
MAXCS Administration Manual iii
C
10
Board Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
Board Configuration Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Using the Triton Resource Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Using the Triton MeetMe Conference Board . . . . . . . . . . . . . . . . . . . . . . . . 109
Configuring the Triton Analog Station Board . . . . . . . . . . . . . . . . . . . . . . . 109
Configuring the Triton Analog Trunk LS/GS and LS Boards . . . . . . . . . . . . . 110
Configuring the Triton VoIP Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
Configuring the Triton T1/E1 Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Configuring the Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Setting up Channels on the Triton T1/E1 Board. . . . . . . . . . . . . . . . . . . . 115
Installing a Channel Service Unit (CSU) . . . . . . . . . . . . . . . . . . . . . . . . . 122
Troubleshooting T1/E1 Common Symptoms . . . . . . . . . . . . . . . . . . . . . . 123
Configuring Virtual Board SIPSP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
Configuring the SIPSP Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
Block Unauthorized SIP Invite Messages . . . . . . . . . . . . . . . . . . . . . . . . 125
Configuring Virtual Board HMCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
Media Pass-Through Support for HMCP . . . . . . . . . . . . . . . . . . . . . . . . . 129
Assign HMCP Resources to IP Extensions . . . . . . . . . . . . . . . . . . . . . . . . 129
HMCP Codec Preference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
Codec Preference - Incoming Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Codec Preference - Outgoing Calls, Third-Party IP Phones . . . . . . . . . . . . 131
Codec Preference - Outgoing Calls, IP Phones. . . . . . . . . . . . . . . . . . . . . 132
Configuring the MAX1000/2000 Board . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Configuring the Virtual MobileExtSP Board . . . . . . . . . . . . . . . . . . . . . . . . 134
C
11
Trunk Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
Opening the Trunk Configuration Window . . . . . . . . . . . . . . . . . . . . . . . . . 136
Selecting Trunks to Set Attributes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
Configuring One or Multiple Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
Setting General Trunk Attributes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
Configuring SIP Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
SIP Group Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
Adding SIP Servers to a SIP Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
Removing Servers from a SIP Group . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
Changing the Order of Servers in a SIP Group . . . . . . . . . . . . . . . . . . . . 145
SIP Server Registration Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
SIP Server General Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
SIP Server SIP Options Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
iv MAXCS Administration Manual
SIP Trunk TLS Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
Caller ID and DID Incoming Sequence Example . . . . . . . . . . . . . . . . . . . . 153
Triton Analog Trunk GS/LS Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
Performing Impedance Match on Your Own . . . . . . . . . . . . . . . . . . . . . . . 157
Using the Match Impedance Button. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Measuring the Rx Level of a Trunk Channel . . . . . . . . . . . . . . . . . . . . . . . 158
If You Need to Improve the Rx Level. . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
If You Don’t Have the Milli-Watt Test Number . . . . . . . . . . . . . . . . . . . . . 159
C
12
In Call Routing Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . .163
Adding and Deleting Caller ID Route Entries . . . . . . . . . . . . . . . . . . . . . . 164
Defining Caller ID Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Adding and Deleting DNIS Route Entries . . . . . . . . . . . . . . . . . . . . . . . . . 166
Defining DNIS Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
C
13
Out Call Routing Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . .169
Configuring Out Call Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
About Route Definitions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Setting Default Routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Configuration Example – Solving 10-digit Dialing . . . . . . . . . . . . . . . . . . . . 174
Resolving Dialing Delay: Non-USA/Canada Countries . . . . . . . . . . . . . . . . . 176
C
14
Extension Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .179
About the Apply To Button. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
Setting Personal Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
Call Recording Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
Physical Location and Type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
Setting the Line Properties. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
IP Extension Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Phone Display Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Configuring Group Options for an Extension . . . . . . . . . . . . . . . . . . . . . . . . 188
Adding or Removing Group Assignments . . . . . . . . . . . . . . . . . . . . . . . . . . 189
Setting up Station Speed Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
MAXCS Administration Manual v
Editing Speed Dial Entries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Setting an Information-Only Mailbox . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
Assign Exchange Integration License . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
SMTP/POP3 Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
Mail Forwarding Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
Setting Message Playback Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
Setting Mailbox Capacities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Setting Message Notification Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Setting the Message Types for Notification . . . . . . . . . . . . . . . . . . . . . . . 194
Emergency Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Unusual VM Activity Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Setting the Type of Notification. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
Setting Notification Timing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
Setting Notification Business Hours . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Enabling Message Notification. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Configuring Calling Restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Setting Call Restriction Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Setting Other Call Restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Setting Answering Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 199
Free Format Forwarding. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Handling Busy Calls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
Setting Call Waiting Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
Handling Unanswered Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
Configuring One Number Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
One Number Access Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
Setting Caller ID Verification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
Specifying Forwarding Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
Restrictions and Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
Configuring a Monitor List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
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15
Setting Up IP Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
Setting VoIP Codec for IP Extension . . . . . . . . . . . . . . . . . . . . . . . . . . . 215
Setting Fax-Over-IP for an IP Extension. . . . . . . . . . . . . . . . . . . . . . . . . 216
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IP Phone Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
Environments with both Polycom and AltiGen IP Phones . . . . . . . . . . . . . . . 217
Configuring the AltiGen IP Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
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AltiGen IP Phone Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 219
General IP Phone Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
Configuring Programmable Keys and Workgroup Status . . . . . . . . . . . . . . 221
Configuring Auto-Discovery of Server IP Address . . . . . . . . . . . . . . . . . . . . 223
Setting Up DHCP Option 120 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
On the AltiGen IP Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Disabling Auto-Discovery. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
When You Have Two AltiGen Servers in the Same Network . . . . . . . . . . . . 227
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Mobile Extension Configuration . . . . . . . . . . . . . . . . . . . . . . . . . .229
MobileExtSP Board Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Configuring the MobileExtSP Board . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
Configuring an Extension as a Mobile Extension . . . . . . . . . . . . . . . . . . . . . 233
Additional Configuration for MaxMobile Communicator . . . . . . . . . . . . . . . 235
Voice Mail for Mobile Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236
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Hunt Group Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .237
Overview of Huntgroup Configuration Window . . . . . . . . . . . . . . . . . . . . . . 238
Establishing Basic Hunt Group Attributes . . . . . . . . . . . . . . . . . . . . . . . . . 239
Establishing Hunt Group Membership . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
Adding Extensions to a Hunt Group. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
Removing Extensions from a Hunt Group: . . . . . . . . . . . . . . . . . . . . . . . . 241
Setting Login Status for System Restart . . . . . . . . . . . . . . . . . . . . . . . . . 241
Setting Hunt Group Mail Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241
Setting E-mail Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 242
Setting Mailbox Playback Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Setting Mailbox Capacities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Setting Message Notification Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Setting the Message Types for Notification. . . . . . . . . . . . . . . . . . . . . . . . 244
Setting the Type of Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
Setting Notification Timing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 245
Setting Notification Business Hours. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Handling Unanswered Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
Setting a Hunt Group’s Call Distribution Rule . . . . . . . . . . . . . . . . . . . . . . 248
Setting Queue Management Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 248
About Fax-over-IP Hunt Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249
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Paging Group Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Setting Up a Paging Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
Adding Members to a Paging Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Removing Members from a Paging Group . . . . . . . . . . . . . . . . . . . . . . . . . 253
Polycom Paging Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Configure Polycom Paging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
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Line Park Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
Implementation notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
Setting Up a Line Park Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
Deleting a Line Park Group. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260
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Workgroup Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
Creating and Configuring Workgroups . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
Overview of Workgroup Configuration Window . . . . . . . . . . . . . . . . . . . . . 261
Setting Up Workgroups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
Establishing Basic Workgroup Attributes . . . . . . . . . . . . . . . . . . . . . . . . 262
Service Level Threshold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
Workgroup Recording Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 264
Recording Tone Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Establishing Workgroup Membership . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
Log In/Out a Group Member. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Setting Login Status for System Restart. . . . . . . . . . . . . . . . . . . . . . . . . 267
Setting Business Hours . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Operational Limitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
Defining Skills for a Workgroup. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
Setting Rules for Skill Based Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
Skill Based Routing Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Setting Workgroup Mail Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . 272
Setting E-mail Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Setting Mailbox Playback Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
Setting Mailbox Capacities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
Voice Mail Access Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
Setting Message Notification Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
Setting the Message Types for Notification . . . . . . . . . . . . . . . . . . . . . . . 275
Setting the Type of Notification. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
Setting Notification Timing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 276
Setting Notification Business Hours . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
Setting Call Handling Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
Handling Busy Calls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
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Handling Unanswered Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
Number of Rings Before Handling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Setting IntraGroup Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Enable Single Call Handling for Agents . . . . . . . . . . . . . . . . . . . . . . . . . . 280
Handling Calls when Group Members Are RNA/Logged Out . . . . . . . . . . . . 280
Announce Agent Info . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Inter Workgroup Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Queue Management – Basic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
Setting Queue Phrase Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
Queue Announcement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
Expected Wait Time Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Queue Overflow Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Supervisor Queue Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
Queue Management – Advanced . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
Announcement Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
Queue Overflow Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
Application Extension Queue Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
Agent Logout Reason Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
Adding a Transmitted CID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
Editing a Transmitted CID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
Deleting a Transmitted CID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
Callback Processing Logic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
Callback from Queue Licenses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 292
Step 1: Assign Licenses and Record Phrases . . . . . . . . . . . . . . . . . . . . . . 292
Step 2: Configure the Workgroups that will Handle Return Calls. . . . . . . . . 292
Step 3: Configure the Workgroups that will Offer the Callback Option . . . . . 293
Blocking Area Codes from the Callback Option . . . . . . . . . . . . . . . . . . . . . 296
Operational Notes for the Callback Feature . . . . . . . . . . . . . . . . . . . . . . . 297
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Using MeetMe Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .299
MeetMe Conference Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
Setting the MeetMe Conference Extension . . . . . . . . . . . . . . . . . . . . . . . . . 300
MeetMe Conference Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300
Working in the MeetMe Conference Window . . . . . . . . . . . . . . . . . . . . . . . . 301
E-mailing a Meeting Invitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
Modifying the E-mail Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
Starting and Stopping a Meeting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
Continuing a Meeting Beyond Its Duration Time . . . . . . . . . . . . . . . . . . . . 305
What the MeetMe Conference Host Can Do . . . . . . . . . . . . . . . . . . . . . . . . 306
Joining a MeetMe Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
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Location-Based E911 (for Roaming Users) . . . . . . . . . . . . . . . . . 307
About E911 Location IDs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
Designing E911 Location IDs for Your Organization . . . . . . . . . . . . . . . . . . 308
AltiGen IP Phones Supporting E911 Location IDs . . . . . . . . . . . . . . . . . . . . 308
E911 Location ID Configuration Process . . . . . . . . . . . . . . . . . . . . . . . . . . 309
. . . . . . . . . . . . . . . . . . . . . . . . . . . . Creating the E911 Location ID Table 309
Requiring a Password to Change E911 Location ID . . . . . . . . . . . . . . . . . . 311
Assigning E911 Locations on AltiGen IP Phones . . . . . . . . . . . . . . . . . . . . . 312
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Network Configuration Guidelines for VoIP . . . . . . . . . . . . . . . . 315
ISP/Intranet Quality of Service (QoS) . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
Ethernet II Framing Header . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 316
WAN Router Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
Network Configuration Guidelines for AltiGen IP Phones . . . . . . . . . . . . . . . 319
Configuration Guidelines for NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Private Network Configuration Example . . . . . . . . . . . . . . . . . . . . . . . . . 319
VPN Network Configuration Example . . . . . . . . . . . . . . . . . . . . . . . . . . . 321
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Enterprise VoIP Network Management . . . . . . . . . . . . . . . . . . . . 323
Understanding VoIP Bandwidth Requirements . . . . . . . . . . . . . . . . . . . . . . 324
Opening Enterprise Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
Overview of Enterprise Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 326
Changing the Enterprise Manager Password . . . . . . . . . . . . . . . . . . . . . . 327
About the G.722 Codec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 332
Assigning Codec Profiles to IP Addresses . . . . . . . . . . . . . . . . . . . . . . . . . 333
Managing IP Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 334
Defining Your Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
Configuring a Public or Intranet Pipe . . . . . . . . . . . . . . . . . . . . . . . . . . . 336
Configuring MAXCS Behind NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 337
The Multi-site VoIP Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 340
Multi-Domain Enterprise Management Features . . . . . . . . . . . . . . . . . . . 341
Creating a Multi-site VoIP Domain. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 343
Declaring Additional Servers for the VoIP Domain . . . . . . . . . . . . . . . . . . 344
Working with Servers in the Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
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Adding a Server to a VoIP Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 346
Rejoining a Server to the VoIP Domain . . . . . . . . . . . . . . . . . . . . . . . . . . 347
Setting an Alternate Server for AltiGen IP Phones. . . . . . . . . . . . . . . . . . . 347
Managing Domain Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
PSTN Failover When the TCP/IP Network is Down . . . . . . . . . . . . . . . . . . . 350
The Scope of an Extension in the VoIP Domain. . . . . . . . . . . . . . . . . . . . . 350
Changing an Extension’s Scope from Local to Global . . . . . . . . . . . . . . . . . 352
Changing an Extension’s Scope from Global to Local . . . . . . . . . . . . . . . . . 353
Relocating a Global Extension for an IP Phone . . . . . . . . . . . . . . . . . . . . . 353
Redirecting AltiGen IP Phones When a Server Is Down . . . . . . . . . . . . . . . 355
Switch Users Back to the Main Server . . . . . . . . . . . . . . . . . . . . . . . . . . . 358
MaxAgent and MaxCommunicator Switchback Behavior. . . . . . . . . . . . . . . 358
MaxOutlook Switchback Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 358
Configuring Departments in a Multi-site VoIP Domain . . . . . . . . . . . . . . . . . 359
Configuring Global Least Cost Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . 361
When Information May Be Out of Sync . . . . . . . . . . . . . . . . . . . . . . . . . . . 362
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System Report Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . .363
System Summary Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363
Resetting Cumulative Statistics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
SNMP Management Console . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
Configuring MAXCS for SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
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Microsoft Exchange Integration . . . . . . . . . . . . . . . . . . . . . . . . . .369
MAXCS Private Cloud Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 369
When You Install MAXCS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 370
Exchange 2010 Integration Configuration Steps . . . . . . . . . . . . . . . . . . . . . 371
Extra Steps for Bridged Access and Native VM Integration . . . . . . . . . . . . . 374
Configuring UM Settings for Each User . . . . . . . . . . . . . . . . . . . . . . . . . . 377
Configuring Out Calling from UM. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
Configuring MaxAdministrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
Exchange 2013 Integration Configuration Steps . . . . . . . . . . . . . . . . . . . . . 384
Bridged Access and Native Voicemail Integration . . . . . . . . . . . . . . . . . . . 386
Configure UM Settings for Each User. . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
Configuring Out Call Routing from UM . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
When You Create a New Mailbox User . . . . . . . . . . . . . . . . . . . . . . . . . . . . 391
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Fax-over-IP Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Configure an IP Extension for FoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
Enabling this setting changes the configuration for this extension as follows: 398
Configure In Call Routing for FoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
Create a SIP Group for Fax Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
Configure Gateway Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 402
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Tools and Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 403
Backup and Restore Utility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 404
Scheduling Backups. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 405
Restoring Backed up Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 406
MAXCS Admin & Extension Security Checker . . . . . . . . . . . . . . . . . . . . . . . 406
Checking Extension Security. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 407
Start & Stop All AltiGen Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408
Parameters for Trace Collector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 409
Network Log for AltiGen IP Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 411
Work/Hunt Group Converter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 414
Exporting and Importing Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Importing Extensions from a .csv File . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Importing Extensions from the Active Directory . . . . . . . . . . . . . . . . . . . 417
Exporting the Extensions in a MAXCS System. . . . . . . . . . . . . . . . . . . . . 418
AltiGen Custom Phrase Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 419
Creating a New Phrase. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 420
Editing a Phrase Name or Description . . . . . . . . . . . . . . . . . . . . . . . . . . 421
Re-recording Phrases. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 422
HMCP Certification Test Tool . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 422
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Quality of Service (QOS) Configuration . . . . . . . . . . . . . . . . . . . 423
Configure the MAXCS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 423
Configure Each MaxAgent Client System . . . . . . . . . . . . . . . . . . . . . . . . . 427
xii MAXCS Administration Manual
A
A
E1-R2 and E1 ISDN PRI Installations . . . . . . . . . . . . . . . . . . . . . .429
Signaling Values, By Country . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 432
Protocol/B Channel Maintenance Message Settings, By Country . . . . . . . . . 441
A
B
Required Service Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . .443
Service Parameters/Request Information for T1 . . . . . . . . . . . . . . . . . . . . . 443
Technical Information for T1 with Voice. . . . . . . . . . . . . . . . . . . . . . . . . . 443
Type of Registered Services Provided . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
T1 Channel Assignment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
CSU/DSU Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
Service Parameters/Request Information for PRI . . . . . . . . . . . . . . . . . . . . 444
Technical Information for PRI with Voice . . . . . . . . . . . . . . . . . . . . . . . . . 445
PRI Channel Assignment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 445
Service Parameters/Request Information for E1 . . . . . . . . . . . . . . . . . . . . . 445
Technical Information for E1 with Voice. . . . . . . . . . . . . . . . . . . . . . . . . . 446
E1 Channel Assignment. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 446
CSU/DSU Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 446
A
C
Network Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .447
Remote IP Phones Behind NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 449
A
D
Technical Support & Product Repair Services . . . . . . . . . . . . . . . .451
How To Reach AltiGen Technical Support. . . . . . . . . . . . . . . . . . . . . . . . . 451
Technical Training for Administrators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 452
A
E
Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .453
Polycom Phone Cannot Register . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 453
Cannot Make a Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 454
IP Resource Does Not Appear in Current Resource Statistics . . . . . . . . . . . . 454
I
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 455
MAXCS Administration Manual xiii
xiv MAXCS Administration Manual
About This Documentation
This manual is designed for AltiGen Partners, administrators, and technicians who are responsible for configuration and administration of a MAXCS 8.0 system.
Related Publications
Related MAXCS documentation, which you can find on the AltiGen Communications web site: http:/
/maxcs.com/support/current-manuals:
•
MAXCS All Software Solution Deployment Guide
•
MAXCS Upgrade Guidelines
•
MAXCS New Features Guide
•
MaxCommunicator Manual
•
MaxOutlook Manual
•
MaxAgent Manual
•
MaxSupervisor Manual
•
MaxInsight Manual
•
MaxMobile iPhone User Guide and Quick Reference
•
MaxMobile Android User Guide and Quick Reference
•
AltiConsole Manual
•
CDR Manual
•
Advanced Call Router Manual
•
AltiGen IP Phone User Manuals
•
ActiveX Manual
•
AlitReport Manual
•
VRManager Manual
•
Polycom Configuration Guide and User Guides
Additional Release Notes:
•
Sightmax Application Notes
MAXCS Administration Manual 1
2 MAXCS Administration Manual
C
H A P T E R
1
New Features
This section describes features that have been added or enhanced since MAXCS Release 6.0.
Enhancements Included in Release 8.0
The following enhancements were included in MAXCS.
Polycom Enhancements
Refer to the MAXCS Polycom Configuration Guide for full details.
•
Zero Touch Provisioning – The process of provisioning Polycom IP phones has been significantly streamlined. Administrators can now configure an option in DHCP (Dynamic Host Configuration Protocol).
The DHCP service, in conjunction with the AltiGen Polycom phone configuration service, will deploy configurations to Polycom phones automatically.
•
Global Extension Relocation – Users can now relocate their Polycom phones within the Enterprise domain.
To enable this feature, use the Enterprise Manager > User > Resolve tab option Allow user to relocate
a global extension using #17/#27
.
To relocate the phone, users connect the phone and enter the extension number and the voicemail password.
•
User Password Consolidation – To simplify Polycom phone management, AltiGen has merged the various passwords in MAXCS into a single password. Specifically, the provisioning password, the voicemail password, and the SIP Registration have been combined into a single password, which is set to the voicemail password.
•
Default Local Administrator Password – You can change the default Polycom administrator password via a new field on the System > Number Plan tab. This is the password for the Polycom phone itself; users must enter this password to access menus on the phone to change its configuration.
In addition, you can enable a Polycom phone so that it downloads this default password; select PBX >
AltiGen IP Phone Configuration, select the extension, and switch to the Polycom tab. Check the
Enable Password Control
option; the password from the System > Number tab will be copied over.
•
Auto-Generate Digit Map – You can now have MAXCS automatically generate a Polycom digit map, which will be based upon your current dialing plan. You do this via a button on the System > System
Configuration > Number tab. We recommend that you review the auto-generated digit map, to make sure that it meets your system requirements. Modify the map as needed.
MAXCS Administration Manual 3
Chapter 1: New Features
•
Mobile Fallback – You can now enable mobile fallback on Polycom phones, just as you can with AltiGen
IP Phones. When enabled for an extension, if a Polycom phone loses its network connection, it will automatically fall back to a Mobile Extension.
To enable this for an extension, open the PBX > Extension Configuration > General tab, check the
Enable Fallback to Mobile Extension
option, and specify the appropriate mobile extension channel from the list.
•
Firmware Auto-Upgrade ( most VVX models) – MAXCS now supports an automatic firmware upgrade to Polycom VVX phones. The Release 8.0 installation includes a specific Polycom firmware (5.2.2.0501).
All VVX 300/310, 400/410, 500, and 600 model phones will be updated to 5.2.2.0501. This firmware version may be changed in future MAXCS releases.
The Auto-upgrade feature is not supported on VVX 1500 model phones.
There are two separate options that you can use to upgrade VVX firmware. One is for new phones that you have recently acquired and are starting to configure (it can also be used on existing phones that are not associated with an extension on this server, such as when an extension number is changed). The other option is for existing phones that are already associated with an extension
•
Global BLF – You can now set a Global extension as a BLF softkey for Polycom phones, just as you can with AltiGen IP phones. This feature requires a Polycom Advanced Features license.
•
Polycom Expansion Module Support (VVX) – MAXCS now supports the Polycom VVX Expansion
Module, which is a third-party device that is sold by Polycom to work with their model VVX IP phones. The
Expansion model is supported on the following Polycom phones:
•
VVX 300/310
•
VVX 400/410
•
VVX 500/600
The BLF slots that you configure will overflow from the VVX phone onto the expansion module. In other words, the BLF slots will fill up all available slots on the Polycom phone's LCD display first, then will begin filling up the expansion module. This feature requires a MAXCS Polycom Advanced Features license.
•
Call Recording Button (VVX) – With Release 8.0, you can enable a Record softkey for Polycom VVX models. This option is found on the PBX > AltiGen IP Phone > Configuration > Polycom tab.
Once this feature has been enabled, the VVX extension user can tap a Record softkey on their Polycom phone to choose to record the call in progress. A new page opens, with a Record button for users to click.
This feature requires a Polycom Advanced Features license.
•
Transfer and Conference Behavior Enhancements – In prior releases, users with Polycom phones had various limitations performing transfers and conference tasks within MaxAgent and other client applications. In Release 8.0, Polycom users now have additional ways to transfer calls and to join or initiate conferences, using the client applications MaxAgent, MaxOutlook, MaxCommunicator, and
AltiConsole. Refer to the MAXCS Polycom Configuration Guide for details. This feature requires a MAXCS
Polycom Advanced Features
license.
•
Corporate Directory Support / User Directory Dialing (VVX) – Users of Polycom VVX phones can now use a corporate directory to place calls, transfer calls, and perform other basic functions. This feature requires a Polycom Advanced Features license.
•
Client Call Control – The MAXCS Clients are now synchronized with Polycom phones. Regardless whether you used the client or the Polycom phone to place a call on hold, you can retrieve the call from either the client or the phone.
•
Video Call Support – MAXCS supports video calls on VVX500 and VVX600 models (requires a separate
VVX USB camera) and on VVX1500 models.
Video conferences are not supported; only direct peer-to-peer calls are supported.
Video calls are supported in Softswitch Deployments only.
Note that MAXCS does not support video over SIP Tie trunks or SIP Trunks.
4 MAXCS Administration Manual
Enhancements Included in Release 7.5 Patch 1
•
Auto-Answer Changes – In previous releases, agents with Polycom VVX phones could both enable the
Auto-Answer feature as well as set the default voice path (speaker or handset) in a single setting the client's Configuration > Extension page.
Due to the specific way that Polycom handles voice paths, AltiGen has changed how users configure this feature. Configuration now requires two separate steps.
•
Additional Enhancements
•
The Caller ID will now show correctly while the call is ringing; in the previous release, the Caller ID updated only after the call connected.
•
When you clear Call Management feature #33 - do not disturb (in System Configuration > Feature
Profiles), the DND feature will also be disabled in Polycom phones. Polycom phones must be rebooted in order for this change to take effect.
Technology Enhancements
The following technology enhancements were included in MAXCS.
•
HMCP Load Balancing – The HMCP system has been redesigned to perform load balancing on a multicore system, in a round-robin method. This results in a more even distribution of work across all channels, resulting in better more consistent performance.
•
SIP Group Configuration – In previous releases of MAXCS, SIP registration was configured by individual channels. This configuration approach had several disadvantages. To streamline SIP registration, AltiGen has implemented SIP Group configuration. The SIP Signaling Channel Configuration dialog box is the main starting point for all SIP channel configuration.
An additional change is that MAXCS now 8.0 checks for valid licenses for SIP Trunks that have been designated as AltiGen SIP trunks. Be aware that trunks will show as not registered if you start MAXCS
Administrator before MAXCS is fully up, because license information will not be ready until then. However, after 30 to 90 seconds, the trunks should show as ready.
See SIP Groups on page 141 for full details.
•
Callback from Queue – A new feature enables organizations to offer callers the option of receiving a return call instead of waiting on hold in a call queue.
You can customize the specific queue conditions under which a caller is offered the callback option. You can also indicate a daily cutoff time for callbacks, so that returned calls can be completed before that workgroup’s business day ends.
For return calls, you can specify how many attempted calls to place, how long to wait between those attempted return calls, and a caller ID to transmit with the returned calls.
See Callback from Queue on page 291 for full details.
•
Android support – MAXCS now supports Android O/S versions 4.4, 5.0, and 5.1.1.
Enhancements Included in Release 7.5 Patch 1
The following enhancements were included in MAXCS 7.5 Patch 1.
Technology Enhancements
•
Free Format – This feature lets you insert delays and/or to send out additional DTMF digits when forwarding calls to an extension, a hunt group, a workgroup, or an outside number. This option is located on the PBX > Extension Configuration on the Answering tab.
See Free Format Forwarding on page 200 for details.
MAXCS Administration Manual 5
Chapter 1: New Features
•
Verizon SIP trunk support – Two new SIP Trunk options, Custom P-Asserted Identity and Diversion, have been added to the SIP Trunk parameters to support Verizon Caller ID. Custom P-Asserted Identity allows you to insert a header into the SIP packet.
•
SIP Trunk Release Link Tie (SIP Refer) - This feature is implemented to support various AudioCodes devices. It instructs the SIP Trunk provider to release both the inbound and the outbound legs of a transferred call once the transfer has been completed. Refer to the configuration guide for your specific
AudioCodes device for full instructions.
Note: Not all PBX or services providers support SIP Refer or release link tie.
See page 147 for more details.
•
SIP Trunk TLS Support – This feature is implemented to support SIPTrunk.com SIP Trunks. For full details, refer to the appropriate SIP Trunk Configuration guide in the AltiGen Knowledgebase. Also see
the discussion in SIP Trunk TLS Support on page 150.
•
IPTalk Redirect – This feature allows users to redirect a client application to the alternate server if the client loses its connection to the main server.
See IPTalk Redirect on page 356 for details.
SIP Delayed Offer Support – MAXCS now supports SIP Delay Offer on incoming SIP calls.
Polycom Enhancements
Refer to the MAXCS Polycom Configuration Guide for full details.
•
Global Relocation – Users can now relocate their Polycom phones within the Enterprise domain.
•
Polycom Password Consolidation – AltiGen has merged the various passwords for Polycom phones, to make maintenance an easier process.
•
Polycom Secondary Proxy – You can now assign a secondary proxy server that Polycom phones will connect to when the main MAXCS server becomes unavailable, for limited inbound and outbound calls.
When the main MAXCS server is back up, the phones will automatically reconnect to the main server.
•
User Password Consolidation – To simplify Polycom phone management, AltiGen has merged the various passwords in MAXCS into a single pass-word. Specifically, the provisioning password, the voicemail password, and the SIP Registration have been combined into a single password, which is set to the voicemail password.
Enhancements Included in Release 7.5
The MAXCS 7.5 release includes enhancements in the following areas (refer to the MAXCS 7.5 New Features
Guide
or the appropriate chapter or manual for full details):
Polycom Enhancements
These features, along with provisioning instructions, are covered in the MAXCS 7.5 Polycom Configuration
Guide
, which can be found in the AltiGen Knowledgebase.
•
Easier Polycom deployment, centralized administration (Hours Offset, Time Format, Digit Map, TLS and SRTP support), #17 phone device override for the same extension ("#17 requires a SIP-Tie trunk to communicate with the server)
•
New Polycom Advanced Features license unlocks additional functionality: BLF, Line Park, DND synchronization with clients, Polycom group paging (Polycom to Polycom only, broadcast limited to local LAN), Client Auto Answer (Polycom VVX models only), client answering device for outbound and inbound calls. caller ID enhancements for Polycom phones, AltiConsole support for Polycom phones.
6 MAXCS Administration Manual
Enhancements Included in Release 7.5
New Licenses
The following new license is now available.
•
Polycom Advanced Features license (Seat license) ALTI-SIPDEVADDON-PLCM-XX – This license unlocks some advanced features, including:
Do Not Disturb synchronization – Allows Polycom users to synchronize DND status between the phone and MAXCS clients
Busy Lamp Field (BLF) and extension monitoring – Allows Polycom users to view the status of monitored extensions, answer incoming calls for monitored extensions, and pick up ringing calls on monitored extensions
Line Park – Allows Polycom users to line-park a call and pick up a parked call
Group Paging – Allows Polycom users to be members of Polycom paging groups, to send and receive pages in those groups
Device Auto-Answer – Allows Polycom users to set their clients (MaxAgent, MaxCommunicator,
MaxOutlook and AltiConsole) to have the Polycom phone automatically answer incoming calls.
Codec Enhancements
Codec enhancements include the following (see Chapter 25, Enterprise VoIP Network Management on page
•
G.722 codec support
•
Managing codec priorities
•
Managing IP network bandwidth
•
Codec negotiation
•
Codec preferences for incoming and outgoing SIP trunk calls
Media Enhancements
•
Media pass-through (HMCP)
•
T.38 pass-through (Note that FoIP support is best effort and can be dependent on the fax device make/ model – MAXCS fax device configuration guides can be found in the AltiGen Knowledge Base)
–
FoIP extension
–
FoIP Hunt group
–
Fax routing
Enhanced SNMP Trap Monitoring
SNMP enhancements include the following (see Chapter 26, System Report Management on page 363 for
details):
•
SIP Trunk group status
•
Out of license condition
•
Out of codec/SIP trunk channels conditions
•
Abnormal conditions (911, suspicious frauds, malicious SIP)
HMCP Tool Enhancements
HMCP Certification Tool enhancements include the following:
MAXCS Administration Manual 7
Chapter 1: New Features
•
DPC latency measurement
•
Generating disk I/O load
•
Generating CPU load
•
G.722 codec load
Additional Enhancements
Additional enhancements include:
•
•
A checkbox for blocking unauthorized SIP invite requests (see Block Unauthorized SIP Invite
•
A SightMax version field (see the MAXCS 7.5 SightMax Integration Guide for details)
•
Many MAXCS 7.5 clients and other applications have new application icons
Client Enhancements
Client Enhancements include the following features:
•
Handling client calls via some models of Polycom phones (requires an Advanced license) – refer to the
MAXCS 7.5 Polycom Reference
for details.
•
Client auto-answer with some models of Polycom phones (requires an Advanced license) – refer to the
MAXCS 7.5 Polycom Reference
for details.
•
Synchronize only Private Contact folders (MaxOutlook)
Features No Longer Supported in Release 7.5
Please be aware that MAXCS 7.5 no longer supports the following features:
•
ACC is not longer supported
•
Windows 2003 Server OS for Softswitch is no longer supported
•
TAPI is no longer supported
•
H.323 SIP Tie-Trunk is no longer supported
Important:
Any IP Dialing Table entries that have H.323 set for the Protocol field must be changed to SIP before you upgrade to Release 7.5.
•
The TLS enabled and SRTP disabled settings are no longer supported
•
The Persistent TLS connection for SIP tie-trunk is no longer supported
•
Redundancy is no longer supported
•
Multiple gateway Softswitch is not supported
Additional Changes in Release 7.5
Please note the following additional changes in Release 7.5:
•
AltiGen Phone Group Paging uses a SIP Tie-Trunk channel in this release.
•
If Polycom phones are used, you must select RFC 2833 for the DTMF Transport field in the Codec
Profile assigned to the SIP tie-trunk.
•
The Release-Link-Tie over SIP tie-trunk feature is not supported in this configuration.
•
The provisioning process for Polycom phones is different in Release 7.5. Refer to the Polycom Phone
Configuration Guide for phone setup procedures.
8 MAXCS Administration Manual
Enhancements Included in Release 7.0 Update 1
Enhancements Included in Release 7.0 Update 1
The following enhancements were included in MAXCS 7.0 Update 1.
•
Operating System Support - Because Microsoft is ending support for Windows XP effective in April,
2014, MAXCS stopped supporting that operating system.
•
Location-Based E911 Configuration (For Relocation) - E911 Caller ID (CID) information is sent to
PSAPs (Public Safety Answering Point) when a user calls 911 from an IP phone. A new feature has been added to MAXCS that allows administrators to configure an E911 Location ID for each geographic location. Each AltiGen IP phone can then be assigned an appropriate E911 Location ID.
This feature is not designed for use with stationery phones, only for phones that are moved from one physical location to another.
Configuring E911 Location IDs ensures that emergency calls placed from these IP phones:
•
Will call the appropriate local PSAP, based upon the phone's physical location.
•
Will transmit Caller ID details that will help emergency response teams find the right location.
Refer to the chapter Location-Based E911 (for Roaming Users) on page 307 for full details.
•
Salesforce Integration - Salesforce.com provides a CTI toolkit to telephony-enable its cloud-based CRM application. When enabled, a softphone control panel appears in the web browser, for agent call control.
AltiGen encapsulates the CTI toolkit into a connector for MAXCS Release 7.0 Update 1, to integrate an
AltiGen IP PBX system with Salesforce.com.
A Salesforce Integration Seat license has been added to the License Information and Client Seat License
Management pages.
Refer to the MAXCS Connector for Salesforce.com Application Notes for full details.
•
AltiReport Enhancements - AltiReport uses an email server to send scheduled reports. In MAXCS 7.0
Update 1, AltiReport can now communicate with encrypted email servers such as the Google mail server
(Gmail) or the Hotmail server.
AltiReport detects the requirements of the specified email server automatically; no configuration is needed to indicate whether the email server requires encryption.
When using an encrypted email server, for authentication purposes the Sender Email Address entry must be the same as the SMTP Account Name entry. The remote email server does not allow users to use a different sender email address than the real email account.
Refer to the AltiReport Manual for full details.
•
Client Enhancements - The following client enhancements are introduced in MAXCS 7.0 Update 1:
•
The MAXCS 7.0 Update 1 clients now support Windows 8.1 (64-bit).
•
MaxSupervisor - Support for up to 50 concurrent MaxSupervisor connections, improved data receiving and display performance. In addition, there is a new automatic re-sort interval for Agent View and Agent
Status. Refer to the MaxSupervisor Manual for full details.
•
MaxCommunicator, MaxAgent, and MaxSupervisor - Users can now choose to display either the local time or the server time for voicemail messages. Also, the application icon now stays on the Windows task bar and in the Windows system tray when you minimize these applications.
•
MaxOutlook Support for Outlook 2013 - The MAXCS 7.0 Update 1 release of MaxOutlook supports
Microsoft Outlook 2007, 2010, and 2013
Note: Only the 32-bit version of Microsoft Outlook is supported.
Users who upgrade their Microsoft Outlook application from version 2007/2010 to Outlook 2013 will need to reinstall MaxOutlook.
MAXCS Administration Manual 9
Chapter 1: New Features
•
Move IP Address between Trusted and Malicious Device List - In some rare scenarios, the MAXCS server can capture regular IP address and add them to the Malicious SIP Device list.
You can now more easily and directly move IP addresses from the Trusted list to the Malicious list. Within
MAXCS Administrator, select SIPSP > Board Configuration > Advanced Configuration. Two buttons
•
Advanced Call Router Enhancements - Advanced Call Router (ACR) allows verification of incoming call information against the database. In earlier releases, ACR only confirmed that the specified information existed; it left the transfer target to be configured in the rule settings. This design mandated that every transfer target needed a unique routing rule, which meant that customers had to create numerous rules to handle calls.
In MAXCS 7.0 Update 1, the routing target information can be retrieved from the database record. This means that customers need to create (and maintain) far fewer rules to process calls.
To accomplish this, rules now have an additional Target action, Route by Query Result. Refer to the
Advanced Call Router Manual
for full details and examples.
•
Block Unauthorized SIP Invite Requests - MAXCS 7.0 Update 1 includes a filter to block unauthorized
SIP Invite requests. When the MAXCS server receives a SIP Invite request, if the IP address or the caller source is NOT in one of the following lists, the call will be ignored (dropped):
–
IP address is in the MAXCS registered IP extension list
–
IP address is in the configured SIP Trunk list
–
IP address is in the IP dialing table list
–
IP address is in the SIP Allow IP list
–
The caller is from another AltiGen system
–
The caller is from an AltiGen IP phone
The filter is disabled by default. To enable the filter, add the following registry entry and press VM&SP
Log Dump
under the Diagnostics > Trace menu (or just restart the system).
HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\AltiGen Communications,
Inc.\AltiWare\Service Providers\SIPSP
"BlockSIPUnauthorizedCall"=dword:00000001
Enhancements Included in Release 7.0
The following features were added MAXCS ACM Release 7.0:
•
Support for Windows 7 32-bit OS for MAX1000
•
Support for Hyper-V virtual machine
•
Hard Disk space checking
•
New “Group Queue” option for Group RNA/Logout Handling
•
Regular admin login to access Diagnostic menu and SIP advanced configuration
•
Support for Polycom IP phones SIP Registration password
•
MaxAdministrator GUI enhancements
•
Changed default T1\PRI setting from T1-CAS to PRI-NI2
•
Support for Exchange 2013
•
Sending SIP Option for SIP Trunk Keep-alive
•
Support for 3rd-party (Polycom) IP phones behind NAT
10 MAXCS Administration Manual
Enhancements Included in Release 6.7 Update 1
•
3rd-Party SIP Phone Capacity improvement (SIP Registration Optimization) ? Enable Respond SIP
Option when the device is behind NAT (Windows Registry)
IP Phone Enhancements
•
IP 710/720 2xA3 increase MIC gain by 6db
3rd Party Integration
•
AudioCodes MP1xx FXS support behind NAT
•
EdgeMarc/AudioCodes Remote Survivability and QoS
•
AudioCodes MP1xx remote Survivability
Tools
•
VoIP NAT/Firewall Test Tool
•
The use of the network assessment tools is independent of the MAXCS release.
AltiReport Enhancements
•
Add Logout Reason description and logout duration to Report 1101
•
Add Total Logout duration to Report 2202
•
Add Total Ring Time (RT) and Avg Ring time (Avg RT) to Report 2206
•
Add DNIS description and Filter by MAXCS DNIS option to Report 3101 and 3201
Sightmax
•
SightMax Integration
Miscellaneous
•
Option to disable trace backup during midnight
•
New VRM conversion library for G.723
•
Backup and Restore utility now includes DB\Storage folder
•
Improve TFTP speed for IP phone firmware update
Enhancements Included in Release 6.7 Update 1
The following features were added in MAXCS ACM Release 6.7 Update 1:
Voice Processing New Features and Enhancements
•
HMCP Softswitch Features -- (Support Soft system key -Dongle; support Windows 2008 Telecom
Server R2 SP1; support 64-bit HMCP driver ; separated NetFilter installation
PBX New Features and Enhancements
•
Support Exchange 2010 integration
Client Application New Features and Enhancements
•
MaxCommunicator -- Updated remote extension activity in MaxClient Directory tab; block Caller ID display in MaxClient Monitor tab
MAXCS Administration Manual 11
Chapter 1: New Features
•
MaxAgent - Updated remote extension activity in MaxClient Directory tab; block Caller ID display in
MaxClient Monitor tab; added Login and Logout All Workgroups buttons in MaxAgent login screen; added server clock to MaxAgent
•
IPTalk Enhancements -- Support compressed codec (G.723) for IPTalk in Windows 7 OS
•
Add-on application enhancements – Support VRManager installation on Windows 2008 Server; support compressed codec (G.723) for VRManager
•
IP Phone New Features and Enhancements – The SIP versions of IP phone firmware are as follows:
–
IP705 – Firmware v2299
–
IP710 – Firmware v2199
–
IP720 – Firmware v2399
Note: With the release of firmware version 2x99 IP705, IP710 and IP720 IP phones users can now press the Release key to reject an internal or external call ringing on the IP phone.
The ringing will stop and the call will follow the extension’s Ring No Answer (RNA) handling which can be the Extension’s VM or other destinations.
Enhancements Included in Release 6.7
The following features were added in MAXCS ACM Release 6.7:
General Server Features
•
Support Exchange 2010 integration
HMCP Softswitch Features
•
Support Soft system key (Dongle)
•
Support Windows 2008 Telecom Server R2 SP1
•
Support 64-bit HMCP Driver
•
Separated NetFilter installation
Client Application Enhancements
•
Updated Remote Extension activity in MaxClient Directory tab
•
Block caller id display in MaxClient “Monitor” tab
•
Login and Logout All Workgroups buttons in MaxAgent login screen
•
Server clock to MaxAgent
•
Support compressed codec (G.723) for IPTalk in Windows 7 OS
Add-on Application Enhancements
•
Support VRManager installation in Windows 2008 Server
•
Support compressed codec (G.723) for VRManager
Enhancements Included in Release 6.5 Update 1
The following enhancements were added in MAXCS ACM Release 6.7:
PBX Enhancements
•
AltiGen IP phone can automatically discover server IP address
12 MAXCS Administration Manual
Enhancements Included in Release 6.5 Update 1
You can enable the AltiGen IP phone to automatically discover the MAXCS server IP address (instead of the user having to enter it manually) by configuring option 120 in your DHCP server with your MAXCS IP address. The user will only need to enter the extension and password.
In addition to making initial IP phone setup easier, this feature is also helpful when there is a need to migrate MAXCS to a new IP address. The administrator just needs to update the new MAXCS IP address in the DHCP server and then reboot all AltiGen IP phones. The phones will automatically pick up the new MAXCS IP address.
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Extensions can be imported from Active Directory
In addition to importing extensions from a CSV file, this update supports importing extensions from the
Active Directory. This saves significant time by avoiding re-entering extension information.
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Recording alert tone is periodic
In addition to inserting a recording tone at the beginning of a conversation, a periodic recording alert tone is added in this update. It occurs every 15 seconds in half-second bursts of 1400 Hz and is recorded together with the conversation. (Configured in Extension Configuration > General page.)
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A Custom Phrase Manager tool makes managing custom phrases easy
The AltiGen Custom Phrase Manager is a Windows-based application that makes managing custom phrases easy. It displays all custom phrases in a graphical user interface. You can add or delete a phrase by clicking a button. You also can rename an existing phrase to a meaningful name. An AltiGen SDK license is required to use this tool.
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Changes in automatic administrative tasks
In the midnight tasks, reset channel is removed, and MAXCS will no longer reset channels and boards automatically.
The configuration backup option under MAXCS Data Management is now turned on by default.
Client Application Enhancement
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MaxCommunicator, MaxAgent, and MaxOutlook enhanced with ability to:
Dial Using Smart Tags – Phone numbers that appear in Internet Explorer and Microsoft Office programs can be dialed through the active MaxAgent/MaxCommunicator/MaxOutlook by either clicking an icon (in
Internet Explorer) or choosing Dial by MaxClient from a Smart Tag (in Microsoft Office programs).
Requires configuration on the MaxAgent/MaxCommunicator/MaxOutlook’s Configuration > General screen and in each Microsoft Office program.
Dial Using Shortcut Keys – Users can select a phone number from any window, for example, Internet
Explorer, Microsoft Word, Excel, Notepad, and so on, and then dial that phone number by pressing two or three keys they define for this task. Requires configuration on the MaxAgent/MaxCommunicator/MaxOutlook’s Configuration > General screen (Select-n-Dial option).
Dial Automatically – Any phone number users dial using a Smart Tag or the Select-n-Dial method can be dialed automatically, or they can choose to simply have that phone number put in the dialer box. This option is configured on the Configuration > General screen.
Record, save and play a message to a call – A new feature called MaxCall allows a user to hand over an outgoing connected call to the MAXCS system so that when the callee’s voice mail is reached it can play a message the user pre-recorded, for example a marketing campaign script. This frees the user’s extension to make the next call. This feature is available on a new MaxCall tab. An AltiGen MaxCall license is needed for this feature.
OCS versions of MaxCommunicator and MaxAgent integrate with Microsoft Office
Communicator
“MaxCommunicator for OCS” and “MaxAgent for OCS” each function as an embedded program in Microsoft Office Communicator 2007 R2. The two new programs are separate from stand-alone MaxCommunicator and MaxAgent and require their own installation. They work much like the stand-alone clients,
MAXCS Administration Manual 13
Chapter 1: New Features with a few differences.
Only one of these OCS products can be used on a client machine at a time.
AltiConsole change – The “Location” column for an extension is added back in.
MaxMobile Communicator extended
iPhone – AltiGen’s MaxMobile Communicator (MaxMobile) application, installed on an iPhone makes the phone a fully capable office phone extension and serves as a “desktop” call control client, allowing the user to access, configure, and perform most of the company's PBX functions directly from the graphical user interface in MaxMobile Communicator.
Android platform – MaxMobile Communicator on the Android platform supports T-Mobile’s myTouch
(G2) phone and Verizon’s Motorola Droid phone, in addition to the T-Mobile G1 phone.
An AltiGen MaxMobile license is required.
License changes
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MaxCall seat and session licenses added
Enhancements Included in Release 6.5
The following features were added in MAXCS ACM Release 6.5:
SNMP Management
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MAXCS issues alerts, via SNMP, that fall into the following categories:
Server status - Server memory, CPU, or hard disk exceeds defined limits
MAXCS Software Status - Switching service is initialized, stopped, or restarted
PRI Service Status - PRI trunk goes down or reconnects
Gateway Status - A gateway server loses connection or restarts
Multi-site Enterprise Manager status - The master or a member server in the Enterprise VoIP domain goes down or reconnects
IP Phone Server Service Status - IP phone server service goes down or restarts
Voice Mail Service Status - Voice mail service goes down or restarts
CTI Service Status - CTI service (CT Proxy) goes down or restarts
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Softswitch Redundancy Status - Softswitch redundancy switchover occurs
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QoS Enhancement (802.1p/802.1Q)
Supports 802.1p class of service priority and 802.1Q VLAN in the server and AltiGen IP phones.
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Secured VoIP Calls (TLS/SRTP)
Transport Layer Security (TLS) and Secure RTP (SRTP) are implemented to establish secured SIP connections and encrypted conversations to prevent eavesdropping. Secured connection can be configured for AltiGen IP phones and SIP-Tie trunks.
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Enhanced 3rd Party IP Phone Support
Release 6.5 supports standard SIP Hold, Transfer, Call Waiting, and server-side Conference for certified
3rd party SIP phones. This release also supports the Polycom SoundStation IP6000 conference phone.
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Microsoft Exchange 2007 Integration Enhancements
Release 6.5 expands the capability of Exchange 2007 Unified Messaging (UM) with the following new features:
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Enhancements Included in Release 6.5
Option to enable voice mail synchronization in Bridged mode. You can use Bridged mode to access
Exchange UM over SIP only or enable voice mail synchronization with UM at the same time.
Native mode integration enhancements, namely, the ability to
Return a call from Exchange voice mail
“Zero out" of Exchange voice mail greeting to the operator
Click "Play on Phone" option from Outlook 2007 to play the voice mail stored in Exchange
In addition, the AltiGen voicemail greeting is disabled when in Native mode.
PBX Enhancements
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Station conference enhancements
The station conference bridge is released when the number of conference participants is reduced to two.
The call can then be transferred or parked. The conference bridge is freed for other users.
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MeetMe conference enhancements
Supports 120 MeetMe conference members in one bridge when using an HMCP MeetMe conference resource. If the MeetMe conference has more than 30 members, by default all the members are muted.
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SIP Trunk Enhancement
Gives the ability to send the Transmitted Caller ID when the extension user makes a call through a SIP trunk, if the SIP trunk service provider supports it.
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Mobile Extension over SIP trunk
Allows you to deliver Mobile Extension features to a cell phone or a PSTN number over a SIP trunk.
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Ability to assign a Mobile Extension to a different trunk group
For systems with PSTN, SIP, or cell phone gateway as MobileExt trunks, you can assign a MobileExt to use a specific trunk group to save toll charges.
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Import and export an extension list from or to a CSV file
Helps speed up the process of creating extensions and configuring extension settings after a system is configured.
Multi-site VoIP Enterprise Management New Features
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Global extension rerouting over PSTN when WAN connection is down.
When a user dials a global extension number in the Enterprise VoIP domain and the WAN connection is down, the call is automatically rerouted over PSTN to the destination. Enterprise Manager will publish the main PSTN number of each site to all VoIP domain members for PSTN rerouting.
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Redirect an AltiGen IP phone to an alternate server when its home server is down.
An IP phone can be assigned to two MAX Communication Servers in the same VoIP domain; one is the home server and the other one is an alternate server. When the home server is down, the IP phones will register to the alternate server automatically. Thus, the IP phone can still work under the alternate
(backup) server. When the home server is recovered, the administrator can switch IP phones back to the home server from Enterprise Manager. The extension must be a global extension in a VoIP domain.
This feature supports only AltiGen IP phones. Analog phones or 3rd party IP phones are not supported.
Client Application Enhancements
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Line Park Supported on MaxCommunicator and MaxAgent
The administrator can assign Line Park groups to an extension in the Line Park configuration. When configured, the assigned lines will show up on the Line Park tab in MaxCommunicator and MaxAgent clients. The extension user can park a call from the client application and allow other users to pick up the call from either an AltiGen IP phone or a client application.
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Chapter 1: New Features
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MaxMobile Communicator (MaxMobile) for the G1 Phone
AltiGen’s new MaxMobile Communicator application, installed on a G1 phone based on Google’s
Android platform, makes the phone a fully capable office phone extension and serves as a “desktop” call control client, allowing the user to access, configure, and perform most of the company's PBX functions directly from the graphical user interface in MaxMobile Communicator. This includes call handling, call forwarding, extension monitoring, conferencing, conversation recording, directory and contact lookup and dial, and contact editing. An AltiGen MaxMobile license is required.
Enhancements Included in Release 6.0
The following features were added in MAXCS ACM Release 6.0:
General
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Integration with Exchange 2007 Unified Messaging
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New client application, MaxCommunicator, replaces AltiView
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Presence state synchronization with Office Communicator (OCS 2007 client)
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TAPI 3.1 support (for Vista OS)
System Features
The following sections list the key features of the MAXCS system.
IP PBX Features
Account Codes - allows the user to input an account code on each call to track telephone usage in order to bill back to clients or create a record of calls specific to a project and to budget and forecast expenses.
Forced Account Codes force the user to input an account code on each call to track telephone usage.
The administrator can configure which extensions are required to enter an account code, and also configure the option to require an account code for long distance calls and international calls, but not local calls. An administrator also can block the display of the account code table in client applications. Users can be prevented from seeing account codes they don’t need to see.
Automatic Dialing Plan Rules- Administrators can configure a call return rule based on the country in which they reside. Applies to call return from Caller ID, Zoomerang, and making a call from Microsoft
Outlook.
Business Hours Profile - allows for setting morning and afternoon business hours for each day of the week. Multiple business hours can be configured in a system. Also, multiple Business Hours profiles can be assigned to DNIS Routing and Trunk In Call Routing entries.
Busy or Ring No Answer Call Handling - sends calls to voice mail, another extension, or AA if the called extension or group is busy or does not answer.
Call Forwarding and Remote Call Forwarding - sends all calls to another extension, to a workgroup/ hunt group, or to an external telephone number. This allows users to redirect their calls to another location, such as home or a branch office. Call Forwarding can be set up either at the source extension or at the destination extension on the system (Remote Call Forwarding). There is 10-hop limit on forwarded calls.
Call Park and Pick Up (Station) - users can park calls at one station to be picked up at another station.
Up to 50 calls may be parked at one station simultaneously. Calls parked to a group are protected. Only group agents or the person who parked the call can pick it up.
Call Park and Pick Up (System) - users can park calls at the system to be picked up at another station.
An ID is assigned to the call when parked. The user can pick up a parked call by entering a feature code and the Parked ID.
Call Park Ring Back Identification to Operator - when parked calls are not picked up, the operator is
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rung.
Call Restrictions - restricts users from dialing specific long distance area codes and phone numbers.
Reduces the risk of toll fraud.
Caller ID - fully supports the Bellcore Caller ID standard and displays alpha and numeric caller ID and name on a standard analog telephone with a display. Up to 64 characters are transmitted and displayed.
If your local exchange carrier provides enhanced caller ID, such as caller name, this information will also be displayed.
Caller ID Routing - the system administrator can define Caller IDs in a routing table and set different routing options.
Centrex Transfer - allows the user to transfer or forward calls to an external telephone number. Once the transfer is complete, the trunk lines are released.
Conference Call (Station) - the system supports conference calls with up to 6 parties, including the dialing extension. You can speak privately to each person before adding the person to the conference. The conference initiator can mute conference members from MaxCommunicator and MaxAgent.
Conference Call (MeetMe) - multiple parties can call into a pre-scheduled conference bridge to join a conference call. The conference host can mute or drop conference members.
Configurable Phone Display - the system administrator can configure the Caller ID, Name, or DNIS number displayed on a phone set.
Conversation Recording - an extension user can record a conversation to voice mail or, with the appropriate license, to a central folder.
Dialed Digit Translation - allows the administrator to select a single dialed digit that can be assigned to route a call to any destination. First Digit Translator allows the administrator to select a single dialed digit that can be assigned to route a call to any destination. Extension Dialed Digit Translator allows predefined dialed digits by an extension to be translated into a different dialing string. The digit manipulation option allows you to remove or add digits to a number dialed by the extension.
Dial Last Caller - allows user to dial the last caller using #69.
Direct Inward Dial (DID) - allows an incoming trunk call to directly access an extension without IVR intervention.
If your local exchange carrier provides DID service, DID calls will automatically be steered to the appropriate destination.
Directory Name Announcement - the extension user’s directory name will be announced to the caller before the call rings to a phone.
Distinctive Call Waiting Tone - allows three different call waiting tone cadences to distinguish between internal, external, and operator calls.
Distinctive Ringing - allows three different ringing cadences to distinguish between internal, external, and operator calls.
Do Not Disturb - blocks all calls coming into a specific extension and sends them to preprogrammed destinations such as voice mail or the operator.
Extension Activity Display and Greeting - allows users to select from a set of pre-defined or customized activity codes that can be played or displayed when the user is absent. A greeting associated with the activity can be recorded and played to the caller. The activity is displayed if the caller is a MaxCommunicator, MaxAgent, or IP phone user.
Extension Based Feature Profile - the system administrator is able to create an extension feature profile that includes enabling and disabling of extension features.
FSK-based Message Waiting - allows message waiting that is based on frequency-shift keying (FSK), a modulation technique for data transmission.
Hands Free (dial tone mute) Mode - by pressing #82, allows a user to leave handset off-hook or use a headset without having to hear the dial tone.
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Chapter 1: New Features
Hands Free (Intercom) Mode - by pressing #81 while on their speaker phone, users can receive internal calls without having to pick up the handset to answer.
Holiday Routing - routes inbound DNIS and trunk calls on designated holidays to specified destinations.
You can create separate routes for business and non-business hours on half-day holidays. Multiple Holiday Profiles can be configured in a system. Also, multiple Holiday Profiles can be assigned to DNIS
Routing, Caller ID and Trunk In Call Routing entries.
Hop Off Calls over VoIP or T1/PRI Tie Trunks - multiple systems at the same or remote locations can be linked through VoIP or T1/PRI networks. Also, this feature provides toll savings on long distance calls by allowing users to dial a remote system via VoIP or T1/PRI trunk and then the destination phone number through PSTN.
Hunt Group - a group of extensions can be set up to perform call coverage, so that if the first extension is busy, the next extension is hunted until a free extension is found. If all extensions are busy, the incoming call will be queued and listen to background music.
Individual and System Call Pick Up - allows users to answer a ringing telephone from another station.
Intercom Call - by pressing #93 on an analog phone, users can make an intercom call to an AltiTouch
510 or an AltiGen IP phone. If the phone is in idle state, the phone speaker will be turned on, and the voice path is connected. If the target phone is busy, the caller will hear a busy signal. This feature can be enabled or disabled per extension by the administrator.
Line Park - allows for a set of 99 lines to be used as a park pool, where trunk incoming calls can be parked automatically, (by routing/call handling treatment in MaxAdministrator). Park Lines are organized into groups, with up to 99 groups supported. Parked Lines can be assigned to an IP phone programmable key for call pickup. Line Park group has busy queuing and time out transfer options.
Live Call Handling - allows a caller to hear a ringback tone when the extension user is in voice mail, paging, transfer, or conference state. Designed primarily for the operator, the call is shown as “ringing” in
AltiConsole.
Meet-Me Conference - MeetMe conference scheduling, monitoring and control.
Mobile Extension - allows a regular CO-connected PSTN phone, such as a home phone or cell phone, to be used to simulate a PBX office extension. A Mobile Extension has most of the PBX system’s call control and call center features. The trunk property is dynamically changed between regular PSTN trunk and MobileExt trunk. The MobileExt user has the option to press any digit to connect to a call. By pressing ** to end a call, the system will simulate on-hook/off-hook sequence and play a dial tone to the
MobileExt user.
Multi-lingual support - supports multiple sets of system and custom language phrases. Up to 9 different sets of language phrase can be configured. A language preference tag can be assigned to the extension user or selected by the incoming caller. The system plays the specified language when the extension user accesses system features or the external caller reaches a voice mail box.
Multiple Call Waiting with Personalized Greetings - a personal queue that allows users to handle multiple incoming calls by letting callers wait in queue until the user answers the call. This allows users to transfer or park calls before answering the next call in queue. Users may also record and use personalized Initial and Subsequent greetings to be played for callers in queue.
Music on Hold - allows callers to hear music or pre-recorded messages while waiting on hold. Music source can be either from an external audio device connected to a telephony board audio input port, or from a pre-recorded music file played by a VoIP board.
One Number Access - a feature that eliminates “telephone tag” by allowing the caller to find the extension user through preset numbers, according to a designated schedule. Setup is available through the
One Number Access tab of Extension Configuration and/or the MaxCommunicator and MaxAgent client applications. An ONA password is optional. The user can press any key to pick up an ONA call. ONA
Call Screening allows a user to enable a call screening option to ONA, where a caller is prompted to record a caller name to continue ONA.
Operator Off-line - when this feature is enabled, all calls are directed to the AA. When the caller dials 0
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and the operator is not available, the call is routed to the operator mailbox.
Out Call Routing Configuration - allows outgoing calls to be directed to particular trunk routes, based on a configured dialing pattern.
Outside Call Blocking - when this feature is enabled, access to outside lines is temporarily disallowed.
Paging (IP) - allows paging over IP to a group of internal IP phones.
Paging (analog trunk or station port) - allows paging through a Zone paging device connected to an analog trunk or station port.
Paging (Audio-Out Port) - allows paging though a speaker connected to an audio output port.
Personal Call Park and Pick Up - users can park calls at one station to be picked up at another station.
Up to 50 calls may be parked at one station simultaneously. Calls parked to a group are protected. Only group agents or the person who parked the call can pick it up.
Single Call Waiting - allows users to put an existing call on soft hold and take a second call upon hearing a Call Waiting tone. The user can then alternate between the two calls.
SIP Third-Party Devices - allows certified third-party SIP devices (for example, a 3rd party IP phone) to register as an IP extension. Note: A license is required to enable this feature for an extension. (Release
5.2)
Station Log In/Log Out - enables system users to move an extension number from one station to another, or deactivate an extension.
System and Station Speed Dial- allows programming of frequently used telephone numbers for speed dialing. Up to 60 system speed numbers can be programmed. Up to 20 station speed numbers can be programmed for each extension.
System Backup and Restore - allows back up of configuration data and voice mail boxes, based on a configured schedule.
Transfer Caller to AltiGen Voice Mail System - allows user to transfer outside caller into the AltiGen
Voice Mail System by pressing FLASH # 40 while connected to the caller.
Transfer Caller to AA - allows a user to transfer a call to an AA by pressing FLASH #15 and then the 2- or 3-digit AA number.
Virtual Extensions - an extension that is not associated with a physical port, but allows access to the
AltiGen Voice Mail System features and telephone sharing.
Workgroup Call Pickup - allows agent or supervisor to pick up a specific call in queue.
Automatic Call Distribution Features
Automatic call distribution (ACD) features include:
Advanced Queue Management Application - enables advanced queuing options:
One-level AA menu selection from queue
Advanced queue overflow for configuration of overflow conditions and actions
After Hours Handling for Workgroups - a workgroup can be assigned a Business Hours Profile through MaxAdministrator. Also, after hours routing decisions can be configured for each day of the week. When a call is forwarded to this workgroup after hours, the call is routed automatically, based on the routing decision for that day of the week.
Agent Login/Logout - allows huntgroup/ workgroup members to log in and out of a group so that incoming calls bypass the workgroup member (agent) who has logged out and the call is automatically routed to other login agents.
Agent Logout Reason Codes - allows a workgroup member to enter a reason code when signing off.
Up to 20 reason codes may be defined.
Agent Set to Not Ready When RNA - when a workgroup call rings an agent and is not answered, this feature automatically sets the agent state to Not Ready.
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Agent Auto Logout When RNA - when a workgroup call rings an agent and is not answered, this feature automatically sets the agent state to Logout for that particular workgroup.
Call Queuing - places caller in a queue to wait until an ACD group member becomes available.
Call Queue Announcement - before a call enters a workgroup queue, the system announces the expected wait time or call queue length to the caller.
Call to Queue Alert - agents can be alerted via a beep and a screen alert when a call enters the workgroup queue.
Distinctive Ringing for Workgroup Calls - allows workgroup incoming calls to use a different ringing cadence from normal calls.
Inter Call Delays - can be used to set delays before the system sends the next call to an extension after the agent finishes an outbound call or other non-workgroup call activity.
Login/Logout/Keep Login Status on system startup or reboot - all group members can be set to the
“Login” or “Logout” state at system startup or reboot. By default, group members are set to “Keep Login
Status.”
Multiple Queue Announcements - allows each group to have its own set of unique audio announcements. Up to five announcements can be configured for each group. The intervals between announcements can also be configured.
Multiple Workgroup Membership - allows each extension to belong to multiple groups. The system can be configured with a maximum of 64 groups (workgroup/hunt groups/paging groups).
Multiple Workgroup Log In and Log Out - lets group members quickly log in and out of multiple groups. (#54 and #56)
Picking/Transferring Calls from Group Queue - enables an extension to pick any call in queue using
MaxAgent or AltiConsole. MaxSupervisor is also able to transfer a workgroup queued call to any extension, workgroup, AA, voicemail or outside number.
Priority Queuing - allows for calls in queue to be associated with a priority. The call priority can be assigned though Caller ID routing, DNIS routing, AA, or other add-on applications. Call distribution is based on the call priority and queue time. Call priority can be escalated if queue time exceeds a certain limit.
Queue Announcement - before a call is sent to a group queue, expected wait time and call position are announced.
Quit Queue Option - a caller can press “#” or “0” to leave a workgroup queue to transfer to group voice mail, AA, extension, another group, or an operator.
Ready/Not Ready - agent can set state to “ready” (#90) or “not ready” (#91) to inform the system whether the agent is able to receive the next call while logged in to a workgroup.
Real Time Monitoring -
Workgroup’s calls in queue, longest queue time, # of calls exceed service level threshold, and service level
Number of agents in Login, Logout, Idle, Busy, Not Ready, Wrap-up, DND/FWD, or ERROR state.
Workgroup and Agent’s performance summary data output to client applications.
Service Level Threshold - a time value for callers waiting in queues. The performance statistics show when workgroup calls are queued for longer than a prescribed value.
Single/Multiple Call Handling for Workgroups - allows the workgroup administrator to select single or multiple calls handling operation for workgroup agents when holding a workgroup call.
Skill-Based Routing - this feature includes the following capabilities:
Assigning skill level requirement (SKLR) to caller
Assigning skill level (SKL) to agent
Matching caller’s SKLR to agent’s SKL
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Setting skill coverage and escalation rules
Supervisor Silent Listen - allows a workgroup supervisor to silently listen to a call between workgroup agent and caller. Personal calls can also be silently listened to by a workgroup supervisor.
Supervisor Barge In - allows a workgroup supervisor to barge into a call between workgroup agent and caller. Personal calls can also be barged in to by a workgroup supervisor.
Supervisor Coach (Whisper) - allows a workgroup supervisor talk to a workgroup agent without the other party hearing.
Queue Overflow Handling - routes incoming calls to an alternate destination when the queue reaches one of the following conditions:
Calls in queue exceed defined limit
Longest queue time exceeds defined limit
Specified percentage of calls in queue with queue time longer than defined service level threshold
Workgroup activity data logging - in addition to CDR data, the following data are logged to a database during workgroup operation:
Agent activity - Login, Logout, Not-Ready, Wrapup, DND/FWD, Error
Agent’s call summary per workgroup
Agent’s call statistics for all workgroups
Workgroup operation summary
Workgroup Activity Monitoring - allows real-time monitoring of workgroup information — group status, call queue status, details of group queue entries, and agent status. Activity summary is available through a group view window in MaxAdministrator, MaxAgent, and MaxSupervisor.
Workgroup Call Distribution - calls can be distributed to the first available group member, or among group members according to the following options:
Ring First Available Member
Ring Next Available Member
Ring All Available Members
Ring Longest Idle Member
Ring Average Longest Idle Member
Ring Fewest Answered Calls
Ring Shortest Average Talk Time
Skill-Based Routing
Wrapup Time - allows a group member some time in between calls to wrap up on notes, prepare for the next call, or log out of the group. This wrapup time is configurable on a per-agent basis.
Auto Attendant (AA) Features
The AA features provide quick and courteous processing of all incoming calls. An AA can be configured to serve as a primary attendant or as a backup to a receptionist.
AA features include:
Dial By Name - allows a caller who does not know your extension number to spell your name using the telephone key pad. The system will search the Directory and make a match on the name to connect the caller to the intended party’s extension. The caller can match first OR last name when dialing by name.
Data-Directed Routing - allows the routing of calls directed by the caller’s input (digit or text). Third-party applications can be used to route incoming calls based on caller information.
Digit Collection - caller can be prompted to enter numbers, which are then collected and used for routing the call.
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Chapter 1: New Features
Direct Station Transfer - allows the AA to handle all incoming calls instead of being answered by an operator. Callers can dial an extension number to reach a specific station or use the name directory to find an extension number.
Mailbox Access- allows employee to log into voice mail box from AA when calling in from outside.
Multiple AA Support - allows up to 255 auto attendants.
Name Directory Service - allows callers to hear a list of employees and their extension numbers.
Programmable Time-Out Handling - allows the administrator to select the action the system should take if there is no digit dialed by the caller within a specified number of seconds.
Set Call Priority - allows the administrator to assign a priority level to an AA menu.
Set Skill Level Requirement - allows the administrator to assign a skill level requirement to an AA menu.
Web-based Call Processing - allows the AA to accept calls placed over the AltiWeb application.
Voice Mail Features
The Voice Mail System is a message management system that provides the calling and the called parties with enhanced communication features. It allows greater accessibility, faster reply time between parties, and reduces the frustration of telephone tag.
The voice mail system includes the following features:
Configurable voice mail playing order - Administrators can configure users’ voice mailboxes to play the oldest or the newest message first.
Disable a Mailbox - voice mailboxes can be disabled so that callers cannot leave messages.
Future Delivery - allows users to record a message to be delivered at a specific time and date in the future, up to one year in advance.
Information Only Mailbox - a mailbox can be configured to announce customized pre-recorded information when accessed. This mailbox does not allow callers to leave a message, but only to listen to the message announcement (personal greeting) from the mailbox. To repeat the message, callers are instructed to press the # key.
Making a Call from the Voice Mail System - allows users to make a call from within the Voice Mail System by pressing # at the Main Menu and entering the internal extension or external phone number. This is especially useful while traveling where users can respond to all messages and make other calls not associated with a message, all with one call into the Voice Mail System. This can result in significant savings. The use of this privilege is configurable on a per-user basis.
Message Management - receives, sends, forwards, deletes, and replies to messages.
Message Notification - designed to alert you of new voice messages in your mail box by calling an extension, phone or pager number. A notification schedule can be set for business hours, after business hours, at any time or at a specified time. You have an option of being notified of all messages or only urgent messages.
New and Heard Voicemails Announced - Heard voicemails are announced, as well as new and saved voicemails, when users access voicemail.
Personal/Activity Greeting - allows users to record personal and/or activity greetings in their own voice to be played when callers reach their voice mail.
Press “0” Option for Extension in Voice Mail - allows a caller to press “0” while listening to an extension’s greeting. The “0” can be configured by the administrator to forward the user to operator or other destinations.
Priority Delivery - allows caller to set the priority of message delivery such as normal or urgent.
Private Messaging - allows users to leave a private message in their voice mail for the expected caller.
Reminder Calls - are used to remind you of important meetings, things to do or people to call.
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Remote Access - allows users to access the Voice Mail System from outside by dialing into the AA and pressing # to log in; or pressing ### from any internal extension to access any voice mail box.
Return to AA - after leaving a voicemail message and pressing # to send it, incoming trunk callers are prompted with the option to return to AA to try another path or person.
Set Call Forwarding from Voice Mail - users can set up Call Forwarding from within the Voice Mail System. This allows users to set up Call Forwarding while away from the office.
Voice Mail Distribution List - allows the user to use system distribution lists or personal distribution lists for forwarding voice mail. Up to 100 distribution lists can be created. Each distribution list can have up to
64 entries, and any entry can be another distribution list.
Zoomerang - allows users to listen to messages in the Voice Mail System, make a return call to a party who left a message, and then return to the Voice Mail System to continue checking the next messages, all in a single call into the Voice Mail System. If the caller ID information is not captured, the user may enter the “call back” number manually.
Internet Integration Features
Internet integration features include:
Exchange Integration - provides message synchronization between MAXCS and a Microsoft Exchange server on the LAN. This feature allows for dynamic synchronization of mail between the two servers so that deleted messages from one server get automatically deleted in the other server. Similarly, a new message sent to one server is transmitted to the other server. This way, the message can be accessed or deleted from either server. For example, when a voice mail is deleted from MAXCS, it is automatically deleted from the Exchange server too.
In addition, MAXCS supports the Exchange Server 2007 Bridged access to Exchange 2007. In this mode, the AltiGen voice mail provides an option to the user to access Exchange 2007's speech enabled auto attendant. MAXCS also supports Native VM integration with Exchange 2007. In this mode, MAXCS uses Exchange 2007 as "Native VM" storage for a system-wide integration with all the AltiGen extensions as a unified messaging enabled Exchange account.
Mail Forwarding - allows you to forward voice mail to an e-mail address. The destination address can be an IP address such as 100.200.101.201, or a domain name such as altigen.com.
Remote Download of Messages via Internet - allows users who are traveling and/or working at home to download all new voice and e-mail messages in the Voice Mail System Post Office Box via a local internet access line.
System and Administration Features
System and administration features include:
AA Configuration File Export- lets you export your complete AA configuration to an html file.
AA Copy - An AA configuration can be copied, forming the template for a new AA.
Alerting - An announcement can be sent to Voice Mail when the e-mail server disk is full.
“Apply To” Feature - applies changes (only the field that was changed) to multiple extensions, trunks or channels instead of having to change them one at a time.
Call Detail Reporting (CDR) - the system collects and records information on outgoing and incoming phone calls, such as length of call, time of call, number of calls. This data is written to an internal database.
Configurable Emergency Number - For international use, allows the system administrator to set up country-specific emergency numbers.
DNIS Routing Tables - incoming trunk calls can be routed to an AA, extensions, workgroups, hunt groups, and so on, based on DNIS numbers configured in the system administration routing tables.
E-911 Calling Support - allows an administrator to designate a number of trunks (Triton Analog or PRI) for dedicated E-911 use. CAMA trunks are supported by analog trunk ports.
MAXCS Administration Manual 23
Chapter 1: New Features
E-mail and Voice Mail Storage - can be placed on drives other than the system drive.
Emergency (911) Call Notification to Extension/Outside Number - when any extension dials an emergency number, the system can make calls to pre-configured extensions or outside numbers. A system can have more than one emergency notification number configured.
Extension Checker - a tool that checks the security status of every extension in your system.
Extension Password Protection for Application Logins - the system maintains a counter for each extension to track CTI client application login failures. When eight successive failures are reached, the system disables login connection for 1 to 24 hours to prevent password intrusion. Applies to login from
MaxCommunicator, MaxAgent, MaxSupervisor, AltiConsole, CDR Search, and other add-on applications.
Feature Profiles - allows administrators control over user access to system feature codes.
License Assignment - A License menu allows administrators to easily verify and assign licenses.
Log In and Log Out - An administrator can log in and log out a workgroup member from the Workgroup
Configuration window in MaxAdministrator.
Monitor List - lets you configure an extension’s privilege to see other extension’s call activity through
MaxCommunicator or MaxAgent.
Password Security - allows administrators to lock extensions that have been “attacked” with false password attempts and to set default system passwords for newly created or newly assigned extensions.
Out Call Routing Configuration - allows outgoing calls to be directed to particular trunk routes, based on parameters assigned in the Out Call Routing table.
Remote Administration - a version of the MaxAdministrator application that can be installed on a Windows 2000/2003/XP client computer to remotely administer one or more systems.
Transmit Extension Calling ID - each extension can be configured with a calling ID. When an outgoing call is made by this extension through PRI or IP trunks, the calling ID is displayed as the Caller ID to the receiving caller.
Voice over IP Features
VoIP features include:
Bandwidth Control for VoIP Sessions - Each server can configure the maximum concurrent VoIP sessions based on its Internet or intranet bandwidth. This feature is to ensure that voice quality will not be impacted if too many VoIP sessions are connected at the same time.
Codec Profile - Multiple codec profiles with different settings can be created and applied to different locations. Each profile can have a different codec, jitter buffer, and packet length to accommodate different IP connections.
DNIS Name Display and Routing over IP Tie Trunk - allows for DNIS information to be transferred to the other system when routed over IP tie-trunks. DNIS name of matched entry can be displayed at Alti-
Console, MaxCommunicator, MaxAgent, and handset.
Caller ID/Name Sent Over IP Tie Trunk - SIP supports sending the caller’s name, so SIP and H.323 calls may display different caller ID information.
DTMF payload embedded with RTP (RFC 2833) - this feature helps to resolve DTMF tone detection and regeneration when using G.723.1 or G.729 codecs. Low bit rate compression can distort DTMF tones during compression and cause the far end device to not be able to recognize the DTMF digits. RFC 2833 specifies a separate RTP payload format to carry DTMF information to ensure the other side can recognize the tone properly.
Dynamic Jitter Buffer - due to various delays in the IP network, audio packet streams may be delivered late or out of order. The system is able to buffer incoming packets and re-sequence them by maintaining a queue. This queue is adjusted dynamically to accommodate different network environment characteristics.
24 MAXCS Administration Manual
Enhancements Included in Release 6.0
Echo Cancellation - due to bandwidth limitations and device loading, long delays may occur during packet delivery process, which worsens the echo effect voice speech. Echo cancellation is provided to maintain reasonable voice quality.
G.711 Codec - toll quality (64K) digital voice encoding, which guarantees interoperability and better voice quality.
G.723.1 Codec - a dual rate audio encoding standard, which provides near toll quality performance under clean channel conditions.
G.729 A+B Codec - speech data encoding/decoding standard of 8 Kbps.
Global IP Dialing Table - The IP Dialing Table is configured in Enterprise Manager. The IP Dialing Table configuration is used to create location-based routing in the Enterprise.
H.323 Tie-Trunk Support - Ensures backward compatibility to systems using AltiGen’s AltiWare versions prior to 5.1.
IP Extension Auto Failover - when an IP extension is unreachable, the system will automatically fail over to a pre-configured Mobile Extension.
IP Group Paging - allows the use of voice paging to IP phone users in a group.
NAT Configuration for SIP/H.323 - When MAXCS is behind NAT with a private IP address, this feature helps to resolve IP address resolution problems when communicating with an external VoIP device.
Silence Detection and Suppression - when silence suppression is enabled and silence is detected, the system stops sending packets to the other side. The other side does not receive any packets and plays silence.
VoIP Hop-Off Call Support - allows an extension to access a PSTN trunk on the remote system and
“hop off” to dial an outside telephone number. This hop off feature can be enabled or disabled on the remote system. Outcall restrictions for hop off calls are configurable.
SIP Trunk Support - MAXCS enables AltiGen’s system to connect to IP-based trunking service providers via SIP.
SIP NAT Traversal - Allows MAXCS to connect to a remote SIP phone or IPTalk behind NAT without changing the NAT setting at the remote location.
Support for RFC 2833 (DTMF payload embedded with RTP) - Supported in SIP trunks only. This feature helps to resolve DTMF tone detection and regeneration when using G.723.1 or G.729 codec. Basically low bit rate compression will distort DTMF tone during compression. The far end device may not be able to recognize the DTMF digits. RFC 2833 specifies a separate RTP payload format to carry DTMF information to ensure the other side can recognize the tone properly.
Support for both SIP and H.323 Tie Trunk - When setting up a system-to-system VoIP tie trunk, either
SIP or H.323 protocol can be used.
Multi-Site VoIP Management - Enterprise Manager
Multi-site management through Enterprise Manager includes:
VoIP domain - when networking multiple AltiGen systems from different sites, one system can be assigned as VoIP domain controller to propagate configuration data to member systems.
Directory Synchronization - when a new extension is added to one of the member systems and configured as Global extension, the VoIP domain controller will propagate this extension to all member systems. Every member system within the VoIP domain will be able to see the extension number plan of other systems.
Multi-site Call Routing - when a user dials an extension number that is not a local extension number, the system will search the Domain extension list. If a list is found, the system will dial the number by using the IP address and extension number stored in the Domain extension list.
Domain User Management - The VoIP domain controller can resolve the conflict if duplicated extension numbers are created in different member systems. This feature also manages extension relocation.
MAXCS Administration Manual 25
Chapter 1: New Features
When an extension user is relocated to another member system, its voice mail and greeting can be moved along with it.
Global Least Cost Routing - when multiple systems are in different area codes or countries, the administrator can set up Global Least Cost Routing to route long distance or international calls through member systems. The routing rules are propagated to all members automatically.
Global Dial-by-Name and Greeting Synchronization - Caller using the dial-by-name feature from any system within the VoIP domain can search the entire global directory. The global extension’s greeting is replicated to all systems within the VoIP domain.
Global Extension Relocation by User - When a global extension user travels to any site, the user can dial #27 to log in to the local server. Enterprise relocates the user’s extension setting and voice mail to the local server and activates the extension as a physical extension. All member systems receive an update notice from Enterprise Manager to change the routing destination.
Global DID Number List - The DID number field is part of the global extension configuration. When a call comes in with a DID number, the system looks for a local extension with the same DID number first.
If the system cannot find a matching local extension, it will match the global extension DID number and route the call.
Global Extension Appearance - With proper configuration, the IP phone user can see the following information for a global extension in the VoIP domain: line state (idle, busy, ring, error), extension status
(DND), and activity (presence). This information can be displayed in MaxCommunicator/MaxAgent/Alti-
Console and on the IP phone. Limitation: For display only; user cannot answer calls for the global extension.
Global Intercom - An extension user can dial #93 + Global Ext. to intercom a Global extension (through a SIP tie trunk.
Optional Add-On Software
The following software is optional:
AltiConsole - a Windows-based Attendant console connected to MAXCS over a network; emulates a standard, hardware-based Attendant console through software; has the flexibility of adding new features through software without changing the hardware.
MaxCommunicator - a Microsoft .NET-based desktop call control and Windows pop-up application that interacts with the system, providing easy-to-use dialing, call control, monitoring, and voice mail management capability.
MaxAgent - a workgroup user version of MaxCommunicator; in addition to MaxCommunicator features, also provides call statistics, call wrap up with data entries, workgroup login/logout with reason codes and agent ready/not ready status.
MaxSupervisor - allows a workgroup supervisor to view an agent’s real-time activity, log in/log out an agent, view workgroup and agent operation statistics, listen/barge-in/coach an agent’s conversation.
All workgroups a supervisor is monitoring are displayed in a single view, making it easy to see what’s happening in all groups at once.
A graphical view (trend lines) displays workgroup statistics to help make better staffing decisions.
Supervisors can check workgroup voice mails without needing a separate license or needing to log in as an agent.
In the MAXCS ACM version, the following features are available:
Color coded priority in queue
Change caller’s priority
Record agent’s conversation with indicator
AltiReport - application that can report an agent’s and workgroup’s operation details, including summary, analysis, and charting.
26 MAXCS Administration Manual
Enhancements Included in Release 6.0
Advanced Call Router - a call handling application that matches incoming call data or collected digits against a customer’s CRM record to determine how to route the call. It has the capability to set call priority and caller’s skill level requirement.
CDR Search - a call reporting tool that allows administrators to search CDR files for records that meet selected criteria, and allows workgroup supervisors to get workgroup CDR statistics.
IPTalk - an IP softphone to allow a MaxCommunicator or MaxAgent user to log in to a system as an IP extension. IPTalk supports G.711 and G.723.1 codec only.
MaxInSight - a workgroup performance application that provides call center managers and agents with the ability to track workgroup status and performance data from a wall-mounted LCD panel or from their
PCs. MaxInSight includes the ability to see the following for single or multiple workgroups:
Real-time queue status
Real-time workgroup resource status
Daily operation results
Trends of data over time
SuperQ - a Java-based application designed to queue and distribute calls for call centers with workgroups located in different geographic locations or across multiple AltiGen servers. SuperQ enables call centers to combine teams of workgroups from multiple locations into one virtual team.
VRManager - allows administrators/supervisors to convert, schedule backup/delete, and query recorded files.
SDK Tool Kit - offers a complete set of tools including APIs, documentation and sample programs, to enable a developer to begin programming rapidly and efficiently. It includes a self-installing CD-ROM containing AltiGen SDK software. Session-based licensing is required for both Basic API and APC API interfaces.
MAXCS Administration Manual 27
Chapter 1: New Features
28 MAXCS Administration Manual
C
H A P T E R
2
2
System Requirements and Installation
This release of MAXCS 8.0 supports Softswitch, Private Cloud, and hardware deployments.
Minimum System Requirements
This section lists the operating systems supported by each MAXCS component, as well as system requirements.
MAXCS All-in-One Softswitch
(Software only)
MAXCS (Max1000)
MAXCS (Max2000)
Office 2G/3G
MaxAdministrator
MAXCS 8.0 Requirements
•
Windows 8.1 Professional (64-bit) supported on Hyper-V 6.0 and VMware ESX 5.5 and 6.0
•
Windows 7 Professional (64-bit) SP1
•
Windows Server 2012 R2 (64-bit) supported on VMware ESX
5.5 and 6.0
•
Windows Server 2008 R2 (64-bit) SP1 with 4GB RAM
•
Windows 7 (32-bit) SP1 with 1GB RAM
•
Windows 7 (32-bit) SP1
•
Windows 2008 (32-bit) SP2
•
Windows Server 2008 SP2 (32-bit) with 2GB RAM
Note: The 64-bit version of Windows will not work on Max2000/
Office series chassis.
Operating Systems:
•
Windows Server 2003 with SP3 (32-bit)
•
Windows Server 2008 R2 with SP1 (64-bit)
•
Windows Server 2008 with SP2 (32-bit)
•
Windows Server 2012 R2
•
Windows 7 (32-bit or 64-bit)
•
Windows 8.1 (64-bit)
System Requirements:
•
Monitor with at least 1024x768 resolution
MAXCS Administration Manual 29
Chapter 2: System Requirements and Installation
Enterprise Manager
HMCP
Integration
MAXCS 8.0 Requirements
Operating Systems:
Note: Windows XP is no longer supported.
•
Windows 2012 R2
•
Windows Server 2008 R2 with SP1
•
Windows Server 2008 with SP2
•
Windows 7
System Requirements:
•
1 GB RAM
•
20 GB available hard drive disk space
Note: The installation program will install JAVA JRE 1.8 automatically
Operating Systems:
•
Windows 7 64-bit
•
Windows Server 2008 R2 SP1 (64-bit)
•
Windows Server 2012 R2
Note: Only Intel-based processors are supported
CDR Database Support:
•
Microsoft SQL Server 2008 Express
•
Microsoft SQL Server 2008 R2
•
Microsoft SQL Server 2012 SP1
Note: Running SQL Server in a MAXCS machine is not supported.
Email Server Integration:
•
Microsoft Exchange Server 2007 , 2010, or 2013
Note: To read online Help pages, the system must be running Internet Explorer 6.0 or later.
Virtual Server Requirements:
MAXCS supports the following virtual server environments:
•
VMware ESX 5.5 and 6.0: Allocate 4 Intel cores @ 2GHz each, 4GB memory, and 160GB hard drive
•
Hyper-V version 6.1
MAXCS Client Applications
This section describes the operating systems supported by each individual client, and lists each client’s minimum system requirements.
Operating Systems
System/Processor
Disk Space
RAM
VRM Server Requirements
•
Windows Server 2012 R2
•
Windows Server 2008 R2 with SP1 (64-bit)
•
Windows Server 2008 with SP2 (32-bit)
•
IBM/PC AT compatible system
•
Intel 2 GHz Pentium 4 or equivalent
40 GB
1 GB RAM
30 MAXCS Administration Manual
Minimum System Requirements
Notes
VRManager client
Operating Systems
VRM Server Requirements
•
The installation program will install JAVA JRE 1.8 automatically.
•
Installation requires 1 GB available hard drive disk space.
However, more space is required for CDR storage.
•
Runs on stand-alone system
•
Windows 7 (32-bit and 64-bit)
•
Windows 8.1 (64-bit)
•
Windows 10
Advanced Call Router Requirements
•
Windows Server 2012 R2
•
Windows Server 2008 R2 with SP1 (64-bit)
•
Windows Server 2008 with SP2 (32-bit)
•
Windows 7 (32-bit and 64-bit)
•
IBM/PC AT compatible system
•
Intel 2 GHz Pentium 4 or equivalent
40 GB
ACR requires a 32-bit ODBC
1 GB RAM
System/Processor
Disk Space
ODBC
RAM
Operating Systems
Applications
System/Processor
Disk Space
RAM
Notes
AltiReport Requirements
•
Windows Server 2012 R2
•
Windows Server 2008 R2 with SP1 with IIS (64-bit)
•
Windows Server 2008 with SP2 with IIS (32-bit)
•
JRE 1.8
•
Database with JDBC Driver SQL Server
•
Client system: Internet Explorer 9.0, 10.0, or 11.0
•
IBM/PC AT compatible system
•
Intel 2 GHz Pentium 4 or equivalent
60 GB
1 GB RAM (2GB RAM required if it runs with SQL Server)
•
Tomcat 6.0 will be installed automatically
•
Must run on a stand-alone system
MaxAgent, MaxCommunicator, MaxSupervisor Requirements
Operating Systems
Note: Windows XP is no longer supported.
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 (32-bit or 64-bit)
Third-Party Integration Software
(MaxCommunicator and
AltiGen Clients)
Outlook 2007, 2010, or 2013 (32-bit only; the 64-bit version of
Outlook is not supported)
System/Processor
Disk Space
•
IBM/PC AT compatible system
•
2 GHz CPU or faster
1 GB
MAXCS Administration Manual 31
Chapter 2: System Requirements and Installation
RAM
Notes
System/Processor
.NET Framework
Disk Space
RAM
Notes
MaxAgent, MaxCommunicator, MaxSupervisor Requirements
1 GB RAM
SVGA monitor (1024x768) with 256-color display or better;
Keyboard and mouse
Operating Systems
3rd Party Software
MaxOutlook Requirements
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 SP1 (32-bit or 64-bit)
•
Outlook 2007, 2010, or 2013 (32-bit only)
Note: The 64-bit version of Outlook is not supported
•
IBM/PC AT compatible system
•
2 GHz CPU or faster
Microsoft .NET 3.5 framework or higher
1 GB
1 GB RAM
•
SVGA monitor (1024x768) with 256-color display or better
•
Keyboard and mouse
Operating Systems
System/Processor
Disk Space
RAM
Notes
ActiveX Control for
MaxCommunicator and AltiAgent
AltiAPI (Alticomlib.dll)
AltiConsole Requirements
Note: Windows XP is no longer supported.
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 (32-bit or 64-bit)
•
IBM/PC AT compatible system
•
1 GHz CPU or faster
1 GB
1 GB RAM
•
SVGA monitor (1024x768) with 256-color display or better
•
Keyboard and mouse
AltiGen SDK Requirements
Note: Windows XP is no longer supported.
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 (32-bit or 64-bit)
Note: Windows XP is no longer supported.
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 (32-bit or 64-bit)
•
Windows Server 2012 R2
•
Windows Server 2008 R2 SP1 (64-bit)
•
Windows Server 2008 SP2 (32-bit)
32 MAXCS Administration Manual
Minimum System Requirements
Operating Systems
System/Processor
Disk Space
RAM
Notes iPhone
Android
MaxInsight Requirements
Note: Windows XP is no longer supported.
•
Windows 10
•
Windows 8.1 (64-bit)
•
Windows 7 (32-bit or 64-bit)
•
IBM/PC AT compatible system
•
1 GHz CPU or faster
1 GB
1 GB RAM
•
SVGA monitor (1024x768) with 256-color display or better
•
Keyboard and mouse
MaxMobile Requirements
• iOS 8.3
•
Android Jelly Bean 4.4, 5.0, and 5.1
Operating Systems
Supported SightMax Version
SightMax Integration Requirements
•
Windows 2012 R2
•
Windows 2008 R2 SP1
•
Windows 2008 (32-bit)
•
Windows 7 (32-bit and 64-bit)
•
Version 1.1.13.358
For Polycom requirements, refer to the MAXCS 8.0 Polycom Configuration Guide.
Boards Supported in MAXCS
The following boards will be supported in MAXCS 8.0 release:
•
Triton T1/E1/PRI Board (2.048 MHz) all revisions
•
Triton IP Boards revision A3 or above, which includes
•
IP 12-Port Board (G.711/G.723.1/G.729A+B)
•
IP 30-Port Board (G.711 Only) (same hardware as 12-Port IP Board)
•
Triton Analog 12-Extension Board Rev A3 or above
•
Triton Analog 12-Trunk LS/GS Board Rev A3 or above
•
Triton Analog 12-Trunk LS Board Rev A3 or above
•
Triton Analog 8 port trunk all revisions
•
Triton Resource Board Rev A3 or above
•
Triton MeetMe Conference board all revisions
•
Triton 0x12 Rev. B all revisions
•
Proton Analog 8-Trunk Board all revisions
MAX 1000 and MAX 2000 Hardware Boards
•
0x12 analog extension board
•
4x8 analog board
MAXCS Administration Manual 33
Chapter 2: System Requirements and Installation
•
8x4 analog board
•
T1/PRI 4x4 combo board
•
T1/PRI board
MAX 4000iG
•
Dual T1/E1/PRI with 60 IP (G.711 only) resources
MAX 4000 All-in-One
•
Dual T1/E1/PRI with 60 combo (G.711/G.723/G.729) resources
•
Analog 4 trunks by 4 extension with 20 combo (G.711/G.723/G.729) resources
•
Resource Board:
•
120 Station Conference/10 Supervisor/30 participant MeetMe (10 bridges)/10 G.711
•
60 Station Conference/30 Supervisor/30 participant MeetMe (10 bridges)/10 G.711
MAXCS Licenses
In MAXCS, most client licenses are available in both concurrent session mode and seat-based mode. Both types can be mixed in a MAXCS system.
The following licenses are available for MAXCS 8.0:
Licenses for MAXCS 8.0
Advanced Call Router
AltiReport
MAXCS ACM
Multilingual
VRManager
Per System
Per system
Per system
Per system
Per system
Per system
Per Seat or Per Session
Per session
Per seat or per session
AltiConsole
Callback from Queue Licenses
Concurrent Recording Session
License
Per session
Dedicated Recording Seat License
Enterprise Manager
IPTalk
MaxAgent
MaxAgent Concurrent License
MaxCommunicator
MaxInSight
MaxMobile
MaxSupervisor
Per seat assigned to record to a centralized folder, and per trunk port with recording enabled
Per server license
Per seat or per session
Per seat
Per session
Per seat or per session
Per session
Per seat
Per seat
MaxSupervisor Concurrent License Per session
Polycom Advanced Features License Per seat
34 MAXCS Administration Manual
Preparation for Installation
SDK Connection Session
SightMax
Third-Party SIP Device
Trunk Control APC SDK
Exchange Integration
SIP Trunk
Licenses for MAXCS 8.0
Per session
Per seat
Per seat registering as an IP extension (non-concurrent)
Per session
Other
Per user
Per activated SIP trunk
The following licenses are available for all-in-one (stand alone) systems only:
Station License
ACM Agent Seat
Licenses for Stand Alone Systems
Per activated extension
Per concurrent login
(Single agent logged into multiple WGs will only take one license)
The following licenses are available for Softswitch/HMCP Media Server/Gateway systems only:
Licenses for Softswitch/HMCP Media Server/Gateway Systems
HMCP Media Server License
HMCP G.711/G.723/G.729 Voice Processing
Resources
HMCP MeetMe Conference
HMCP Supervision
Softswitch Station License
Softswitch ACM Agent Session License
Softswitch ACM Agent Migration License
Gateway
Available resources in the system
Available resources in the system
Available resources in the system
Available resources in the system
Per extension configuration
Per session
Per ACC agent session
Per gateway
Preparation for Installation
Before you start installing MAXCS, you need the following:
•
Windows Update – Make sure your server has the recommended Windows Service Pack or Update.
•
MAXCS 8.0 installation media – The MAXCS installation DVD or other media that contains the MAXCS
8.0 programs.
•
MAXCS latest update – Check to see if there is an update available to the MAXCS 8.0 Release.
•
System Key – The system key can be either a USB hardware security device that must be attached to the server MAXCS is running on, or it can be a soft system key.
•
You cannot use both a hardware device and the soft system key at the same time.
•
When using a soft system key, the MAXCS system must be a member of an Active Directory domain.
•
Software license key – A 20-digit key located on the front of the End User License Agreement.
MAXCS Administration Manual 35
Chapter 2: System Requirements and Installation
Installing MAX Communication Server
Softswitch installations – If you are installing the MAXCS 8.0 Softswitch version, follow the instructions in the MAXCS All-Software Solution Deployment Guide.
Upgrades – If you are upgrading from a previous release of MAXCS, review the instructions in the MAXCS
8.0
Upgrade Guidelines before you begin any installation steps. If you are upgrading from a release prior to
MAXCS 7.0 Update 1, then the installation program will check whether Microsoft Outlook has been installed on the server. If it detects Outlook, you must uninstall Outlook before you can install MAXCS 8.0.
In addition, please review the detailed steps in the readme file, which is located on the installation DVD or other media.
To install MAXCS, insert the DVD and follow the instructions on the install screens. when prompted, select a setup type.
•
All-in-one Hardware System Installation – Select this option if you have a hardware system that includes AltiGen boards, such as Max1000, Max2000, and 3G.
•
Softswitch System Installation – Select this option if you have a softswitch system. On the next screen you can select which components to install.
36 MAXCS Administration Manual
Installing MAX Communication Server
Softswitch System Installation
If you chose to install the Softswitch System installation, you can install the following components.
•
Softswitch (including VM and Enterprise Manager) – Select this option to install Softswitch to the server. Softswitch provides the following functions:
–
Devices Control
•
IP Phone
•
HMCP Media Server
•
IP Gateway
–
Call Control
•
Call Signal Processing (SIP tie trunk)
•
PBX Switching, Routing, and Call Handling
–
System Management
•
Configuration and Directory
•
Phrases and Prompts (System, Custom, Personal)
–
Feature Server
•
Voice Mail Server
•
Multi-Site Enterprise Manager
•
Call Center Feature Server
•
CTI Server
•
Exchange Integration Server
•
CDR Server
•
HMCP Media Server – For softswitch deployment, select this option.
The HMCP Media Server check box is not available if the operating system is not supported by HMCP
(refer to the operating systems listed in Minimum System Requirements on page 29). AltiGen supports
HMCP Media Server only on servers provided by AltiGen.
•
Gateway – This option is not supported in Release 8.0.
MAXCS Administration Manual 37
Chapter 2: System Requirements and Installation
For detailed instructions on installing and upgrading MAXCS, AltiGen Partners should refer to the readme file and to the MAXCS 8.0 Upgrade Guidelines.
Installing MAXCS Administrator on a Network Client
MaxAdministrator can be installed on a client workstation, providing the ability to manage the MAXCS server remotely. The system running MaxAdministrator and the MAXCS server must be on the same Windows domain.
When you install MaxAdministrator on a machine that is not a MAXCS server, it does not contain the switching,
SMTP/POP3 server, messaging agent, AltiBackup, and Exchange integration services that are included in the full MAXCS installation. Remote MaxAdministrator does not utilize the System Data Management or Shutdown
Switching Service functions on the MAXCS system.
To install MaxAdministrator on a client workstation,
1.
From the MAXCS installation media, run SETUP.EXE in the MaxAdministrator folder.
2.
Follow the instructions on the screen.
Uninstalling MAXCS 8.0
To uninstall MAXCS 8.0 be sure to stop all MAXCS-related services before uninstallation. To do this, run
MAXCS Administrator, log in, and select Services > Shutdown Switching from the menu.
In the event that the auxiliary services were not stopped, stop them one at a time in Windows, using the Start
> Programs > Administrative Tools > Services tool.
Then go to Start > Programs > Control Panel > Add/Remove Programs, and select MAX Communication
Server ACM 8.0, and then click Remove.
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3
Getting Around MaxAdministrator
This section gives a brief overview of MaxAdministrator, the program used to configure and administer MAXCS and its client applications.
MaxAdministrator has a graphical user interface with tabbed windows that makes it easy to use.
Note: The commands Services > Utilities > Backup and Restore, and Services > Shut Down All
Services cannot be performed remotely.
Logging In and Out
To log into MaxAdministrator,
1.
From the Windows Start menu, select All Programs > Max Communication Server ACM >
MaxAdministrator 8.0.
Figure 1. Log into MaxAdministrator
2.
Enter your username and password.
Important:
The first time you log in, use the system default password, 22222. To ensure system security, change the system password as soon as possible.
To log out, click the Logout button, or select Services > Logout.
Changing the Password
Select Services > Change Password to change to a new password.
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Chapter 3: Getting Around MaxAdministrator
The Main Window
When you run MAXCS Administrator, you’ll see a window similar to the following figure:
Menu bar
Figure 2. MaxAdministrator main window
The main menu bar is at the top. Below that are buttons for quick access to more commonly used configuration screens. A status bar at the bottom contains information on the current runtime status.
The Main Menu
These are the menus and the functions found under each menu:
•
Services – Log in and log out, change password, access utilities (backup and restore, convert work/hunt group, import and export an extension list), shut down all services, and exit.
•
System – Opens windows where you can configure system settings, gateways and media servers, voice mail, auto attendants, multilingual support, call recording, application extensions. Also where you can request certificates and create directories for Polycom users (refer to the MAXCS Polycom Configuration
Guide
).
•
PBX – Opens windows where you can configure trunks, in call routing, out call routing, extensions, AltiGen
IP phones, hunt groups, line park, and MeetMe conference. You can also manage MeetMe conferences from this menu.
•
Call Center – Opens windows where you can configure workgroups, agent logout reasons, and MaxCall.
•
VoIP – Opens windows where you can configure the enterprise network, the multi-site domain, and the refresh enterprise settings.
•
Report – Opens windows where you can view the system summary and IP traffic statistics and configure
SNMP (simple network management protocol).
40 MAXCS Administration Manual
The Main Window
•
Diagnostic – Opens windows where you can view the trace, view the system log, and view the AltiGen network log.
•
View – Lets you show, hide, and set default alignment of the view windows, the toolbar, and the status bar.
•
Help – Opens the Help window and offers a link to the AltiGen Technical Support site.
Quick Access Toolbar
Toolbar buttons give you quick access to frequently used functions.
Figure 3. MaxAdministrator quick access toolbar
From left to right, the toolbar buttons serve the following purposes:
Login. Opens the Password dialog box to log in to the system.
Logout. Logs out of the system.
System. Opens the System Configuration window, or the System menu.
Shortcut for System > System Configuration.
Trunk. Opens the Trunk Configuration window.
Shortcut for PBX > Trunk Configuration.
Extension. Opens the Extension Configuration window.
Shortcut for PBX > Extension Configuration.
Workgroup. Opens the Workgroup Configuration window.
Shortcut for CallCenter > Workgroup Configuration.
IP Phone. Opens the IP Phone Configuration window.
Shortcut for PBX > AltiGen IP Phone Configuration.
AA. Opens the AA Configuration window.
Shortcut for System > AA Configuration.
Recording. Opens the Recording Configuration window.
Shortcut for System > Recording Configuration.
MeetMe. Opens the MeetMe Conference window.
Shortcut for PBX > MeetMe Conference.
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Chapter 3: Getting Around MaxAdministrator
Summary. Opens the System Summary window.
Shortcut for Report > System Summary.
About. Opens a window that displays version and file information. Gives information about the AltiGen Technical Support Web Site. This is a shortcut for Help > About
MaxAdministrator.
Status Bar
The Status Bar at the bottom of the main window displays disk usage, the status of SMDR, the status of the call detail reporting log, the status of the operator, and current date and time.
The View Windows
The MaxAdministrator main window hosts a number of child windows that provide various views into the internal system real-time status.
Boards View Window
The Boards window displays the hardware board types and their logical and physical IDs. For each installed board, it displays:
•
The board’s logical ID (the sequential ID of the board assigned by the system).
•
Board type.
•
The physical ID (including the ID on the faceplate of the board and the gateway ID). If it is an all-in-one system, the gateway ID is the system itself, and the ID is 0.
Double-click a board to open a configuration window for that board.
Figure 4. Boards window
Click on any column heading to sort by that column. Click again to reverse the sort order.
Extension View Window
The Extension View window displays the name, location, and status of all assigned extensions.
Click on any column heading to sort by that column. Click again to reverse the sort order.
Double-click any extension number to open the Extension Configuration window for the selected extension.
The radio button to the left of each extension number is green when the extension is idle, and red when the extension is not ready or in use. The Location number (for example, 01:0005) identifies the card logical ID and port (channel) number on the board. For example, in location 01:0005, the card logical ID is 1 and the port number is 5. If an IP Extension is logged on, the location will show the IP address.
42 MAXCS Administration Manual
Right-click an extension to display its physical location...
The View Windows
...or to open a line properties window specific to the selected extension.
Figure 5. Extension View window
The Reset button resets the selected extension to the idle status. You’ll be asked to confirm the reset.
You can click the Reset button without selecting an extension, and then type in the extension number for the extension to reset.
Trunk View Window
The Trunk View window displays the status of all trunks.
The trunk type or group name shows in the
“Type” column.
Right-click a trunk to display its physical location or to open a properties window specific to the selected trunk.
Figure 6. Trunk View window
The radio button to the left of each trunk location is green when the trunk is idle, and red when the extension is not ready or in use. The location format is logical board ID:channel – for example, channel 3 on the board in logical board ID 9 is location 09:03. The Type, Status, and Duration of trunk use are also shown.
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Chapter 3: Getting Around MaxAdministrator
Note: The Duration field displays the duration of the trunk only if the call is connected after MAXCS
Administrator is started. The field will be empty if the trunk is idle, not ready, out of service, or the call was connected prior to MaxAdministrator being launched.
You can double-click any trunk location to open the Trunk Configuration window for the selected trunk.
The Reset button resets the selected trunks to the idle status if the trunk is connected to a carrier. You’ll be asked to confirm the reset, and a status message will tell you if the reset was successful.
Call Log View Window
The Call Log View window displays the line and trunk traffic history.
Prints selected log entries.
Clears the window of all data.
Figure 7. Call Log View window
The window displays, for the last 30 calls, the caller line or number, the callee, the starting time in 24-hour format and the length of the call. When the call is from another AltiGen system, the call is displayed as “Caller
System IP Address-Extension Number
.”
Workgroup View Window
The Workgroup View window displays data and statistics for workgroups:
Figure 8. Workgroup View window
This window displays the following data:
•
Extension – The workgroup pilot extension number
•
Name – The workgroup name
•
Agents – The number of agents assigned to the workgroup
•
Login – The number of agents logged into the workgroup
•
Available – The number of logged in agents who are available to receive workgroup calls
•
DND – The number of logged in agents who are unavailable with the Do Not Disturb status
•
Wrapup – The number of agents who are in wrapup mode
•
Not Ready – The number of logged in agents who are in Not Ready state
•
Busy – The number of logged in agents who are currently on the phone
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The View Windows
•
Error – The number of logged in agents with extensions that are left off-hook or other user error
•
Logout – The number of agents who are logged out from the workgroup
•
Unstaff – The number of agents who are logged out from the system and have become a virtual extension
•
Queue – The number of calls waiting in queue
•
Waiting Time – The longest wait time of callers in queue
•
Service Level – The percentage of calls in queue with queue time less than or equal to the defined service level threshold
Current Resource Statistics Window
The Current Resource Statistics window displays the total VoIP channels, available channels, and in-use channels.
The window allows administrators to monitor VoIP channel usage and MeetMe conference resource use.
Figure 9. Current Resource Statistics window
Top part of the window
Contains a summary of codec usage.
Middle part of the window
Displays the following data:
•
Gateway ID – The ID of the VoIP channel’s home gateway
•
IP Resource – The Triton VoIP logical board ID:internal DSP channel ID
•
Codecs Capability – The codecs the IP channel can use
•
Active Codec – The codec currently being used
•
Used By – The extension, trunk, or SIP channel that is using this channel
•
Connect To – The extension, trunk, or channel the channel is connected to
•
Packets Sent/Received – The number of voice packets sent and received
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Chapter 3: Getting Around MaxAdministrator
•
Bytes Sent/Received – The total size (in bytes) of all voice packets sent and received
•
Network Packet Loss – The number of voice packets that have been lost due to prolonged delays, network congestion, or routing failure
•
JB Packet Loss – The number of voice packets that have been discarded due to jitter buffer overflow
•
Total Packet Loss Rate – The ratio of total number of lost packets versus total received packets
•
Max Packet Loss Rate – The maximum packet loss rate observed over a period of time during a whole session
•
Jitter – Displays the average length of delay per voice packet in milliseconds. This number can be used to measure the quality of service on the network that connects the source and destination sites. Under 100
on page 327 for more detailed information on jitter.)
•
Local Ports – Displays the local RTP/RTCP port for the voice stream
•
Remote IP Address:Port – Displays the remote RTP port for the voice stream
Bottom part of the window
Shows information about the MeetMe 30-party conference bridge:
•
Gateway ID of the 30-party conference board (for example, 00)
•
MeetMe Conference Bridge ID (from 00:00 to 00:09)
•
Number of members currently participating in a conference using each bridge
Note: Each system can have only one 30-party MeetMe conference board.
Setting the Refresh Interval
The Current Resource Statistics window is updated according to the Refresh Interval configuration. By default, the Refresh Interval is set to refresh the data in the window every 5 seconds. To change the refresh interval, click the Refresh Interval button at the top of the window, and set the refresh interval to a number of seconds up to one minute. To set the time to 0 is to turn off the refresh interval.
Assigning Seat-Based Client Licenses
Most MAXCS client products require either session- or seat-based licenses. You may have purchased both types. A session license allows a certain number of extensions to use a client product.
If you have purchased seat licenses so that particular extensions always have access to the client product, those extensions must be assigned to the client product in MaxAdministrator. If an extension is not assigned to a product, that extension may not be able to use the client product. You may have seat-based licenses for the following client products:
• MaxCommunicator
• MaxOutlook (uses MaxCommunicator license)
• MaxAgent
• MaxSupervisor
• MaxMobile Communicator
• AltiConsole
• IP Talk
• Integration Connector
• MaxCall
• Callback from Queue
Assign extensions to seat-based licenses in the Client SEAT License Management configuration screen
(License > Client SEAT License Management). Licenses are sorted in alphabetical order.
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Stopping the AltiGen Switching Service
Figure 10. The Client SEAT License window
Select a license type and then select extensions to add to the list of “members” who can always use the selected product. To make multiple selections, use Shift+click and Ctrl+click. The screen shows the total number of licenses you have for a client product and the number of licenses assigned.
Stopping the AltiGen Switching Service
Normally, when you exit MaxAdministrator, the AltiGen services that provide the various telephony and data services remain active. If you need to shut down the phone system, do one of the following:
•
From MaxAdministrator, select Services > Shut Down All Services.
•
From Windows, select Start > All Programs > MAX Communication Server ACM > Utilities > Start
and Stop All AltiGen Services, and click the Shutdown All AltiGen Services button.
This stops the MAXCS system services, including the MaxAdministrator application itself. When you re-open
MaxAdministrator, the switching services are reactivated.
These options are available when you are logged in at the MAXCS system computer; they are not available from a remote MAXCS client.
Note: Stopping the AltiGen services from the Windows Services tool is not recommended, because it requires you to know what all the services are and is time-consuming.
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Chapter 3: Getting Around MaxAdministrator
Programs Available from the Windows Start Menu
Several MAXCS programs are available from the Windows Start menu.
Figure 11. The programs on the Start > MAX Communication Server ACM menu
Available under MAX Communication Server ACM:
•
MaxAdministrator – Lets you configure and administer your MAXCS system.
•
Readme – Readme file for MAXCS ACM.
Available under Gateway Tools:
•
AltiGen Board Test – A hardware test tool used to debug system hang and other hardware problems.
See “AltiGen Board Test” on page 403.
•
•
Gateway Configuration – Lets you view some gateway settings and board information and change the
ID and password of a gateway.
Available under HMCP Tools:
•
HMCP Configuration
Available under Utilities:
•
ACM Backup and Restore – Backs up your configurations and extension voice mail. See “Backup and
•
MAXCS Admin and Extension Security Checker – Checks the security status of every extension in your
MAXCS system. See “MAXCS Admin & Extension Security Checker” on page 406.
•
Read Config – Creates a subdirectory of HTML files that shows details of your MAXCS configuration. See
•
Start and Stop All AltiGen Services – Opens a dialog box that gives you the option to start or stop all
AltiGen services by clicking a button.
•
Trace Collector – Collects the trace in selected MAXCS categories, within a time range specified, for
debugging purposes. See “Trace Collector” on page 409.
•
Voice File Converter – A voice phrase conversion tool that converts WAV files to ADPCM, WAV to PCM,
or ADPCM/PCM to WAV format. See “Voice File Converter” on page 412.
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4
System Configuration
The System Configuration window is where you can configure the MAXCS system-wide settings.
To open this window, select System > System Configuration.
The System
Configuration button
You can then work with the following settings, each of which is accessed by a tab in the System Configuration window.
•
General setup – System ID, area code and number, operator and manager extensions, country, distinctive ring, conference call, and system call park options
•
Number Plan – How the system responds to each first digit dialed
•
Business Hours – Used by system functions
•
Holiday – How calls are routed on designated holidays
•
System Speed – Speed dial numbers that can be used by all extension users
•
Call Restriction – Prefixes to block, toll call prefixes, and call control
•
Account Code – Tables for creating and removing account codes
•
Call Reports – CDR logging and data export
•
Country Relevant – Settings for toll call prefixes and emergency numbers
•
Audio Peripheral – Settings for music on hold and system default prompts
•
Activity – Settings for pre-defined or customized activity codes
•
Feature Profiles – Settings for extension feature profiles
Setting General Parameters
Use the General tab in the System Configuration window to set the system ID, area code, main number, and country; extensions for the manager, the operator, the MeetMe conference administrator; and options for distinctive ring, conference bridge, and system call park.
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Chapter 4: System Configuration
Figure 12. System Configuration, General tab
You can set the following parameters and options:
System General
Parameter
System ID
Country
Manager Extension
Description
Assign a number (1-100) to the system. This ID will be used to differentiate call records if multiple systems are writing call records to a same external database.
In a multi-site VoIP domain, each System ID/Server ID must be unique and must be the same length. Once a server is part of a VoIP domain, you cannot change the System ID.
Select a country for the system. This configuration ties to a tone table matched to the country's telecom interface specification.
Select the system manager’s extension number.
The system manager has access to the following system administrator functions:
•
Record custom phrases
•
Turn on trunk blocking (#38)
•
Manage voice mail’s System Distribution List from phone
•
Run CDR search as administrator login account
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Setting General Parameters
System General
Parameter
Distinctive Ring
System Home Area Code
Conference Bridge Option
System Main Number
PRI Calling Number
System Call Park
•
Timeout, Ring Back in ...
Minutes
•
Play Greeting Phrase
Operator Extension and Group
Members
Call Supervision
Description
Enables users to distinguish between internal, external, and operator calls by the way the phone rings:
•
Enable Distinctive Ring – establishes a short double ring cadence for internal calls and a normal, single ring for external calls. When disabled, both rings are normal.
•
Enable Operator Priority Ringing – produces a long single ring between short pauses on calls to the operator.
•
Enable Workgroup Priority Ringing – produces a short single ring between short pauses on calls to the workgroup.
Note: This feature is not supported on Polycom phones.
Area code for the system location.
Note: This field cannot be blank in the U.S. and Canada.
Selected, conference calls will end when all internal lines have disconnected from the conference bridge.
Not selected, the conference connection can continue between outside parties, even after all internal parties have disconnected.
The main system telephone number, which is sent to the pager’s display when a user’s messaging options are configured to call a pager.
This number will be used by a PRI trunk as the outbound caller ID in the event that no number is assigned in the trunk Phone Number, 10digit DID, or extension Transmitted CID field.
Note: This field cannot be blank.
Check the check box to send a caller’s caller ID when the call is going through one-number access (ONA) or when the call is being forwarded.
System Call Park (#41) allows the extension user to park a call, then pick up the call from another extension. If the call is forgotten, the
Timeout sets the number of minutes a call remains parked before the user’s extension rings again. To the caller, the call park sounds like being put on hold. Valid entry: 1 – 60 minutes.
Select a greeting that the caller will hear before being placed on hold.
Select the extension to be used by the system operator. If the extension number you select is a workgroup or a hunt group, member extensions will show up in the Group Members box.
This extension is used in the following applications:
•
Trunk incall routing
•
DNIS incall routing
•
Auto Attendant
Check the box to allow supervisors to monitor, barge in on, coach, and record an agent’s non-workgroup calls.
Notes:
•
If this check box is checked, the supervisor can listen, barge-in on, coach, and record an agent's conversation regardless of the agent's login status.
•
Supervisor extension does not have to be a workgroup member to listen to, barge-in on, coach, or record an agent's conversation.
•
For the coaching feature, the agent’s extension can be either an IP extension or a Triton analog extension.
WARNING! Listening in to or recording a conversation without the consent of one or both parties may be a violation of local, state and federal privacy laws. It is the responsibility of the users of this feature to assure they are in compliance with all applicable laws.
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Chapter 4: System Configuration
Setting a System Number Plan
The system number plan defines the extension digit length. You can use from 3 - 6 digits for extensions. You also use the system number plan to set a DID number length to use, and to define the system response to the first digit dialed — for example, pressing 9 to get a trunk line.
The numbering scheme requires some thoughtful planning.
To set the number plan, select System > System Configuration, then click the Number Plan tab.
Figure 13. System Configuration, Number Plan tab
52 MAXCS Administration Manual
Setting a System Number Plan
Use the Number Plan tab to specify the following parameters:
System Number
Plan Parameter
Extension Number
Length
Default Password
DID Number Length
Dialed Digit Translator
Description
The number of digits for your extension numbering system. Valid entries are from
3 - 6. For example, extension 2001 and 4020 are
4-digit extension numbers.
Note: Once the first extension is configured, the extension number length cannot
be changed
without totally reconfiguring the system or deleting all the extensions already configured.
Further, if a first digit dialed is assigned to extensions and you have set up extensions beginning with that digit, you cannot change the digit assignment without first deleting all affected extensions. For example, if 7 is assigned to Extension and you’re using extensions 7010, 7113, and so on, you cannot reassign 7 to IP trunk access, without first deleting all the
7nnn extensions.
The default password for newly created extensions is randomly generated by the system. (When the password is changed, it must be four to eight digits in length.)
The number of digits needed to match a DID (Direct Inward Dialing) number. The range is from 2 – 16.
Each extension can be assigned a DID number. A DID number does not have a fixed length. For example, suppose the DID number length is 4 and the extension
DID number is 2522999. Depending on the service contract with the Central Office
(CO), the DID trunk can send all 7 digits (2522999) or just the last 4 digits (2999).
If the DID Number Length option is set to 4, the system always tries to match the last 4 digits received to the last 4 digits of a DID number, regardless of what is received.
Note: To accommodate future growth and minimize disturbance, it is recommended that the length of the DID numbers assigned to an extension be greater than or equal to this DID Number Length.
This feature is capable of intercepting and manipulating a dialed digit string before it is sent out for outbound call processing.
To set up a dialed digit translator entry, check the Enable checkbox and click the
Setup key. This opens a dialog box where you can select First Digit Translator or Extension Dialed Digit Translator.
This feature supersedes the first digit assignment of the system number plan.
When configured, any extension user can dial a single DTMF digit that will be translated to any internal or external number. After digit manipulation, the translated digits go through the system number plan to find the internal or external target. For example, you can configure “*” to call an internal workgroup to report an urgent situation.
Typical applications are:
•
One-digit emergency dialing
•
One-digit dialing to branch or headquarters over PSTN or VoIP
•
One-digit dialing to activate a feature code
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Chapter 4: System Configuration
System Number
Plan Parameter
Description
First Digit Translator Configuration
Figure 14. Single Digit Routing
To set up a First Digit Translator entry, select the check box (to the left of 1-9, * or #), then enter the desired digits. When a box is checked, the digit preprocessor will replace the first digit 1-9, * or # that user dials with the digits indicated in the corresponding field. In the above example, if a user dials “*”, the system replaces this with “911”.
Note: This feature is for internal extension users only. It does not support dialing out from voice mail. Improper configuration may cause conflict with the system numbering plan. Be sure to fully test any configuration change in this area before going “live.”
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Setting a System Number Plan
System Number
Plan Parameter
Description
Extension Dialed Digit Translator
Note: This feature is intended for, but not limited to, allowing a remote IP extension to make an emergency call (911) through MAXCS. If MAXCS is in a different location than the IP extension, the emergency call can be routed to the emergency center where the IP extension is located.
Figure 15. Extension Call Routing
To set up an Extension Dialed Digit Translator entry:
1.
Select Extension Dialed Digit Translator from the Select Digit Translator list.
2.
In the Extensions Group field, use the Add button to create and select an extension group that the Extension Dialed Digit Translator will apply to.
3.
(optional) From the Non members list, you may select an IP extension that the
Extension Dialed Digit Translator will apply to. You can apply the same
Members to multiple locations. You may also enable the Bypass Account
Code option if Account Codes are required.
4.
Enter digits in the Dialed Number field and Translate To field. In (see
Figure 15), assuming the system is located in area code 510, when an IP ex-
tension user in LA dials “911,” MAXCS will translate the digits into
“919495550911.” (9 = IP trunk access code, 19495550911 = the emergency center in LA that covers the remote IP phone user’s area.)
5.
The Manipulation option allows you to remove or add digits to a number dialed by the IP extension.
The most common situation requiring this option is to hop-off a VoIP call from a remote system to a remote CO line.
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Chapter 4: System Configuration
System Number
Plan Parameter
First Digit Assignment
Default Polycom Local
Admin Password
Polycom Phone Digit
Map
Description
These define how the system responds to the first digit dialed by the user. The list options for each digit are:
•
Extension
•
Trunk Access
•
Feature Access
•
Operator
•
Invalid (no action)
•
IP Trunk Access
•
Route Access
Trunk Access
– Defines how to get a PSTN trunk line to dial an outside number.
“9” is the default trunk access code.
If you have a more complicated dialing number and routing plan, change “9” to
the Route Access code and configure the Outcall Routing table.
Feature Access
– By default, # is set to Feature Access, which is used as part of feature access codes. In addition, you may also set 1-9 to Feature Access. For example, if 7 is set to Feature Access, Station Login (#27) can also be accessed using 727.
IP Trunk Access –
Only one IP trunk access option is allowed per system. To use
Voice over IP, you must set up this access and, in addition, configure the IP Dialing
Table as discussed in “Defining the IP Dialing Table” on page 338 and set the VoIP
codecs as discussed in “Setting VoIP Codec Profiles” on page 327.
Note: After setting the IP Trunk Access code here, you should set the Trunk Access Codes of any 30-port VoIP boards to “None” on the General tab of
boards – which use the G.711 codec only – for calls to MAXCS servers or other gateways that may require the G.723 codec. If you still want users to have access to this trunk for outgoing calls, you can set it up through out
call routing (see Chapter 13, Out Call Routing Configuration).
Route Access –
The Route Access option can be assigned to one or more digits, to route the call per the out call routing table. Out call routing, which is sometimes called ARS (Automatic Route Selection) or LCR (Least Cost Routing without
carrier rate table), is described in Chapter 13, Out Call Routing Configuration.
Out call routing is designed to help 10-digit dialing, Zoomerang dialing, digit manipulation, and tie trunk hop-off dialing.
You can change the default Polycom administrator password (which is currently
456). This is the password for the Polycom phone itself; users must enter this password on the phone in order to access menus on the phone to change its configuration.
The system will initially generate a random 5-digit string; you can change this to a string between 4 and 32 characters.
See the MAXCS 8.0 Polycom Configuration Guide for details.
See the MAXCS 8.0 Polycom Configuration Guide for details.
Setting Business Hours
The Business Hours tab contains group boxes for setting the business hours and days of the week for which the business or organization is in operation. The business hours schedules are used to set other system settings such as trunk, and DNIS and caller ID in-call routing.
56 MAXCS Administration Manual
Setting Business Hours
Note: Because the business hours are used throughout the system, you or the appropriate administrator must make sure the system time has been set correctly. The system time can be changed using the Date and Time options in the Windows Control Panel.
To access the Business Hours settings, select System > System Configuration, then click the Business
Hours tab.
Figure 16. System Configuration, Business Hours tab
Multiple Business Hours profiles can be configured in a system. A default “System” Business Hours profile is already configured. Multiple Business Hours profiles can also be assigned to DNIS Routing and Trunk In Call
Routing entries.
To add a Business Hours profile, click Add. Enter a name for the profile, then click OK.
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Chapter 4: System Configuration
For each business hour profile, set the business schedule parameters.
System Business Hours
Parameter
Day
AM and PM Schedules
Description
Select the days of the week on which the company does business. For example, if the company does business Monday – Friday, check the check boxes for those days.
For each day of the week, select the time periods during which the company is available for business. The time between the AM and PM times can be used to indicate a lunch break or time between shifts.
If you don’t want to set a break between AM and PM schedules, set the
PM starting time to be the same as the AM ending time.
Or if you want to specify 24 hours as standard business hours, select the following hours:
AM Schedules: From 08:00 AM to 12:00 PM
PM Schedules: From 12:00 PM to 08:00 AM
Routing Calls on Holidays
You can create special routes for incoming DNIS and trunk calls that come in on designated holidays. For holidays that your organization treats as half-days, you can create separate profiles for business and nonbusiness hours.
Note: Incoming DID and tie trunk calls will not follow holiday routes, but go to the dialed extensions directly.
To configure Holiday routing rules, select System > System Configuration, and then click the Holiday tab.
Figure 17. System Configuration, Holiday tab
Multiple Holiday Profiles can be configured in a system. Each Holiday Profile can include multiple holidays. A default “System” Holiday profile is already configured. Multiple Holiday Profiles can also be assigned to DNIS
Routing and Trunk In Call Routing entries.
58 MAXCS Administration Manual
Configuring System Speed Dialing
Creating a Holiday Profile
1.
Click the Add button beside Profile. Enter a name for the profile, then click OK.
2.
To each profile, add holidays that will be included in that profile: Click the Add button below the Holiday list to create a new holiday.
3.
In the dialog box, select a date from the calendar and enter a description. Click OK.
The holiday you added appears in the Holiday list. Additional holidays you create appear in the list and together make up the Holiday Profile.
Setting Call Routing
1.
Select a Holiday Profile from the Profile list, and then select a holiday in that profile from the Holiday list.
2.
Set call routing for “normal” holiday hours using the field group in the Normal section of the Holiday tab.
This will be the default route for calls coming in on that holiday.
3.
If you have special work hours during holidays, check the Special hours option and configure special hour routing.
This route will override the route for normal holiday hours, for the hours you specify. Use this option, for example, to route calls for the working portion of a holiday that your organization treats as a half-day.
4.
To apply these hours to more than one holiday, click the Apply To button and in the Apply To dialog box, select all the holidays to which you want the hours to apply. You can select multiple holidays by using Ctrl-
click or Shift-click. Click OK.
5.
When you are finished with the dialog box, click OK.
When a new year begins, the dates on which holidays fall usually change. You can edit the dates for annual holidays, making them accurate for the new year.
Updating the Dates of Annual Holidays
1.
Select a Holiday Profile, and then the holiday from the Holiday list. Its date and description appear in the
Normal section.
2.
Click the drop-down arrow beside the date to open a calendar and assign a new date. Click Apply.
Configuring System Speed Dialing
You can set up to 60 system speed dial numbers. The IDs available are from 00 – 59. Users press #88, and follow that with one of the system speed dial access codes you set here.
To configure Speed Dialing, select System > System Configuration, and then click the System Speed tab.
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Figure 18. System Configuration, System Speed tab
Adding Speed Dial Entries
To add a speed dial entry,
1.
Click Add.
2.
The next available ID is filled in for you, or you can select the ID number using the drop-down arrow.
3.
Type a name for the Speed Dial entry, then enter the full number as you would dial it, with a maximum of
20 digits per entry. For example, the phone number 914085551212 comprises 9 (trunk access code), 1
(long distance prefix), followed by 408 (area code), and then the seven-digit telephone number.
Valid digits include 0 through 9, #, *, and (,) comma. The comma represents a one-second pause, when
IP trunks are not used.
Editing Speed Dial Entries
To edit an entry, double-click the number you want to work with, or select the number and click Edit. In the dialog box that opens, edit the entry and click OK.
Figure 19. Speed Dial Configuration
To delete a system speed dial entry, select it in the System Speed tab and click Delete.
Note: System speed dial is read-only from MAXCS and MaxAgent.
Defining System Call Restrictions
The Call Restriction tab contains settings for the following functions:
•
Block calls to area codes from all extensions
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Defining System Call Restrictions
•
Define local/toll-free (unrestricted) area codes
•
Lock an attacked extension
•
Block all outgoing trunk calls
•
Restrict other system users from hopping-off to make an outbound call via a tie trunk
•
Set 10-digit dialing area codes for using trunk access code
To set up call restrictions, select System > System Configuration, then click the Call Restriction tab.
Figure 20. System Configuration, Call Restriction tab
Blocking Calls to Area Codes from All Extensions
To add or edit system-prohibited area codes:
1.
Double-click an index entry in System Prohibited Prefixes list, or select the index entry and click Edit.
This opens a dialog box that allows you to enter a prefix number.
2.
Enter a 1 and the dialing prefix to block (for example, 900, 976). You can enter up to 20 digits maximum for each prefix. For example, to block calls from all extensions to 976 numbers, type 1976. Click Apply.
Note: A maximum of 20 prefixes can be defined.
Setting Unrestricted Area Codes
To add or remove “local” call definitions (including calls that begin with 1 but are free: 800, 888), use the Add or Delete button in the Unrestricted Area Code panel, and click Apply. The Extension Configuration’s
Restriction tab references these area codes (as local and unrestricted) in its Outcall Restrictions panel.
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Locking Attacked Extensions
If a user enters eight consecutive invalid passwords when logging on to voice mail or to activate an extension,
MAXCS considers this an attack. To protect your company from theft of services, you can lock an attacked extension for the period of time you specify (10 minutes – 23 hours, 59 minutes, and 59 seconds) in the
Password Check field group.
Extension Security Checker” on page 406.
Blocking All Outgoing Calls
To block all outgoing calls – for example, during the night when no employee is in the office – check the Block
All Outgoing Calls check box. If you block all outgoing calls, then calls between two systems will not go through. This is because #38 blocks all trunks, including remote SIP-tie trunks
Enabling Hop Off for Tie Trunks
When selected, this function allows users from another system to borrow a PSTN trunk in this system to make an outbound call over a T1 or VoIP tie trunk.
Outgoing call on
PSTN Trunk
System A
Tie Trunk:
T1/PRI or
VoIP
System B
The call originating on System B hops off over the T1/PRI or VoIP trunk to use the PSTN trunk in System A. System A must be set up to allow System B to borrow its PSTN trunk.
Figure 21. Hop Off for Tie Trunks
Restricting Tie Trunk Calls
You can set call restrictions on hop-off calls by telling the system to use the same restrictions as the ones set up for an extension. Using the Call restriction follows extension list, you can select the extension with the restrictions to use for the hop-off calls.
Setting 10-Digit Dialing Area Codes
The 10-Digit Dialing Area Code field lets you define area codes that do not require dialing a “1” before the area code. To enter an area code, click the Add button.
Note: Applies only to calls that use a trunk access code. For calls using a route access code, 10-digit dialing area codes must be configured in the Out Call Routing Configuration window, Dialing Pattern tab. See
“About Dialing Patterns” on page 173.
Creating Account Codes
Account Codes let you enable or force users to assign incoming and outgoing calls to particular account codes for billing, tracking, or forecasting purposes. Up to 10,000 account codes can be created.
To access the Account Code tab, select System > System Configuration, then click the Account Code tab.
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Setting up Call Reports
Figure 22. System Configuration, Account Code tab
Adding and Deleting Account Codes
To create an account/code association, click Add. Enter an Account Name and Account Code in the dialog box. The Account Code may contain 1-10 digits.
To delete an account and its code, select it and click Delete. You can select multiple items for deletion by using
Ctrl-click or Shift-click. Click Apply to save your changes and OK to save and close the window.
You can now set options for each extension that determine whether account codes must be entered or can be bypassed. You can also block display of the Account Code table (in which case, you would want to supply
users with the account codes they need). See “Setting Personal Information” on page 181.
Setting up Call Reports
You can set up the call report logging option only if MAXCS and MaxAdministrator are installed on the same server.
On the Call Reports tab, specify the following:
•
Where to log the call detail records (CDR). The location can be an internal database, an external database, or both.
•
How you want the system to manage an internal CDR database.
•
If CDR needs to be output through a COM port to another computer, which COM port and which baud rate to use.
To learn more about internal and external CDR databases and schema, refer to the CDR Search Manual.
To set up Call Reports, select System > System Configuration, then click the Call Reports tab.
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Figure 23. System Configuration, Call Reports tab
Internal Database Configuration (Internal Log Service)
The Internal Log Service (shown in the Log Service display table) is created by default. You can enable or disable the service, but you cannot remove this database or add another Internal Log Service.
To manage the internal CDR database:
1.
Make sure the Internal Log Service check box is checked.
2.
In the Internal Database Configuration field, use the up/down arrows to select the Active database
retaining period in months. This determines how long the data will be kept in the database. Valid entry is 1-12 months.
3.
(Optional) In the Archive purged record(s) field, use the up/down arrows to select the number of months per archive file. This determines the number of months that the system will archive an existing CDR database before creating a new database.
4.
Press OK or Apply.
External (Remote) Logging of Call Data
you need to set up and configure the SQL database and external logger application. Please refer to the CDR
Search Manual
to learn how to set up an external logger service.
Some considerations:
•
The SQL database cannot be on the same server as the MAXCS system. A system integrator or database developer will need to write a custom query to extract data from the SQL database.
•
You can send reports from a number of different systems to the same database.
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Country-Relevant Settings
•
AltiGen does not provide any SQL backup and restore utility. We strongly recommend that you use SQL
Backup and Maintenance utility to perform daily backup and maintenance jobs, and use a restore utility to restore the database. If you need to reconstruct the SQL server, run the External Logger Setup to create an empty calldb database before restore.
•
There is no AltiGen license required for external logging.
To set up and enable external CDR login service to the local or network drive, click Add.
Figure 24. Add External Log Service
Fill in the fields, and click OK.
Name
Address
Port
Password
Parameter Description
The name of the external log service machine (optional)
The IP address of the external log service machine
The TCP port of the machine
The password to connect to the external service machine
Exporting Through a Local Port
You can send the CDR to a COM Port to export to, for example, a call accounting data processing system.
To do this, select the Enable Data Output box in the Accounting Data Processing field group. Then select an Available Port and the Baud Rate.
Country-Relevant Settings
The Country Relevant tab in the System Configuration window contains group boxes for setting toll call prefixes and emergency numbers.
The Country field displays the country selected on the System Configuration, General tab.
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If your system is not in North America,
The Automatic
Dialing Plan Rules button is available.
Figure 25. System Configuration, Country Relevant tab
Setting Toll Call Prefixes
MAXCS uses Toll Call Prefixes to determine the type of outside call and imposes restrictions when necessary. For example, if the international toll call prefix is 011 and a user attempts to make an international call from an extension without international call privileges, the call will be terminated as soon as the user dials 011 after the trunk or route access number. The caller hears an error tone.
•
Domestic. The dialing plan for your country’s domestic long distance prefix. For example, type in a 1 for
1-plus dialing within the U.S. dialing plan (also known as the North American Numbering Plan).
•
International. The prefix used for international calls. For example, this is 011 for international calls made in the U.S.
Setting Emergency Numbers
The number in the Emergency Number field will have the system automatically find a trunk to process the call without the extension user dialing a trunk access code first. You may enter up to three emergency numbers in the appropriate fields.
Note: This feature works with both trunk access code and route access code.
Dialing Plan Rules for Non-North American Country
If your MAXCS system is in a country other than the U.S.A. or Canada, you can configure a call return rule based on the country, which will greatly improve the call return feature from Caller ID, Zoomerang, and making a call from Microsoft Outlook.
Click the Automatic Dialing Plan Rules button. The following dialog box opens:
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Audio Peripheral Configuration
Figure 26. Automatic Dialing Plan Rules dialog box
Define the Local Plan, Domestic Plan, and International Plan. A character of the pattern can be a digit from 0 to 9. It can also be a range of digits, for example, [0-3]. If it is a question mark, '?', it is equivalent to [0-9].
When return calls are made, these rules are followed:
•
When the number matches Local Plan, the system will send the number out to the trunk directly.
•
When the number matches the Domestic Plan, the system will send the number out with the domestic toll call prefix.
•
When the number matches the International Plan, the system will send the number out with the international toll call prefix.
When a number matches multiple entries, the match with the most digits has priority.
Audio Peripheral Configuration
You can configure audio peripheral settings:
•
Music on hold
•
System default beginning and update prompts for callers in queue
To access the Audio Peripheral configuration window, select System > System Configuration, then click the Audio Peripheral tab.
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Figure 27. System Configuration, Audio Peripheral tab
Configuring Music On Hold and Recorded Announcements
Callers will hear the music or recorded announcement configured on this tab only if the user places the caller on hold.
To configure music on hold when using audio equipment,
1.
Check Enable Callers on Hold or in Queue to Listen to Music or Recorded Announcement.
2.
Select the Triton Analog Station board number to which the audio equipment is attached.
To configure music on hold to play a file,
1.
Make sure a VoIP board is installed (required for playing a file).
2.
Check Enable Callers on Hold or in Queue to Listen to Music or Recorded Announcement.
3.
Use the list to select the logical board ID of the VoIP board.
The system will play the default music-on-hold file when the user places the caller on hold.
The default music-on-hold file is a .wav file called “MusicOnWaiting.wav”. The file is located in the
C:\PostOffice\phrases\Music folder. You can replace the file with a .wav file (or an AltiGen PCM file). The .wav file must be in 8 kHz/ 8 bit/ Mono/ u-Law format. Any optional music-on-hold files included with MAXCS are in that format.
Note: You may need to reduce the music volume level 70-80% to avoid distortion.
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Activity
To replace the default music-on-hold file,
1.
Back up the default file.
2.
On the Audio Peripheral tab, uncheck the Enable Callers on Hold or in Queue to Listen to Music or
Recorded Announcement check box.
3.
Rename the desired .wav file to “MusicOnWaiting.wav” and put it in the C:\PostOffice\phrases\Music folder.
4.
On the Audio Peripheral tab, check the Enable Callers on Hold or in Queue to Listen to Music or
Recorded Announcement check box.
Note: If you have two files named MusicOnWaiting in the MusicOnWaiting folder, one a .wav file and one a
PCM file, the .wav file takes precedence.
RTP Resource Usage
In the event that MAXCS is controlling multiple gateway systems, the music source can come from the primary system or another gateway system. When a music source is in one gateway and listeners are in another gateway, one VoIP resource channel in each gateway is used to convey the music stream.
Setting Greeting and Update Prompts
To play a prompt before placing the caller into a hold queue,
1.
Select the Play Prompt Before Placing the Caller in Queue check box.
2.
Use the list to select the prompt number you want to use for the greeting message. (Creating prompts is
discussed in “Phrase Management” on page 85.)
To play an update prompt every 60 seconds,
1.
Check the Play Update Prompt Every 60 Seconds check box.
2.
Use the list to select the prompt number you want to use for the greeting message.
Note: These settings will be used by all hunt groups and workgroups as the default system queue phrase.
However, these settings will be overridden by the workgroup’s queue management phrase setting.
Configuring Overhead Paging
Note: This feature is not supported in the MAXCS Private Cloud service.
1.
Connect overhead paging equipment to the audio out jack on a Triton telephony board.
2.
On the System Configuration > Audio Peripheral tab, select Enable Overhead Paging.
Use the list to select the board to which the overhead paging is attached.
Activity
The Activity configuration tab is used to configure activity codes that can be displayed at AltiConsole when the extension user is absent. MaxCommunicator users, MaxAgent users and AltiGen IP phone users can select from these activity codes to let others know where they are when they are away from their desks
(meeting, business travel, and so on).
A greeting associated with the activity can be recorded and played to the caller. When the user changes the
Activity, the extension’s greeting is also automatically changed to the greeting associated with this activity.
To access Activity configuration, select System > System Configuration, then click the Activity tab.
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There are a total of nine activity codes; the first six are pre-configured as follows:
1 – System Default, 2 – Personal, 3 – Meeting, 4 – Away From Desk,
5 – Business Travel, 6 – Personal Time Off
The remaining three activity codes (7, 8, 9) are not assigned and can be customized by the administrator. To customize an activity code, click on the activity code and click Edit.
In the Edit Activity dialog box, enter name of the Activity and click OK.
Feature Profiles
Select System > System Configuration, then click the Feature Profiles tab to configure feature profiles.
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Feature Profiles
Figure 28. System Configuration, Feature Profiles tab
The Feature Profiles configuration tab allows the system administrator to create an extension feature profile that includes enabling or disabling of the following extension features:
AltiGen Voice Mail:
## <pwd> – Login to VM
#14 – Personal Options
Making Calls:
#34 – Dial by Name
#35 – Station Privilege Override
#93 – Intercom
Answering Calls:
#29 – Individual Call Pickup
#30 – System Call Pickup
#31 – Personal Call Park
#41 – System Call Park
#51 – Line Park Pickup
#81 – Hands Free Mode *
#82 – Dial Tone Disabled *
Call Management:
#17 – Polycom Phone Device Override (#17 requires a SIP-Tie trunk to communicate with the server)
#26 – Station Logout *
#27 – Station Login *
#32 – Enter Account Code
#33 – Do Not Disturb **
#36 – Call Forwarding
#37 – Remote Call Forwarding
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Other Features:
#12 – Language Setting *
#38 – Outside Call Blocking
#39 – Operator Offline
#44 – Overhead Paging *
#45 – Overhead Paging by Trunk *
#46 – Group Paging *
#53 – Outgoing Workgroup
#54 – Login Workgroup
#56 – Logout Workgroup
#59 – Workgroup Call Monitor
#66 – Trace Collection
#73 – Silent System Call Park
#90 – READY to Receive Workgroup Call
#91 – NOT READY to Receive Workgroup Call
* Feature is not supported by Polycom IP phones
** For Polycom IP phones, the phone must be rebooted before the change will take effect
Note: If the extension is an IP extension, #26 / #27 is still available when the phone is in the onhook position, even if it is disabled in the extension’s feature profile.
Adding Feature Profiles
By default, a System feature profile is assigned as 0. To add a new Feature Profile, click the Add button and type a name for the feature profile.
Note: When adding a feature profile, the system will automatically assign the lowest available number.
Select the check boxes for the MAXCS feature codes that you want to be associated with this feature profile, then click Apply.
After the System Administrator creates a Feature Profile, the Feature Profile can be assigned to a specific
181 for more information on assigning a feature profile to an extension.)
Important:
If you assign a feature profile (for example: 2 – Sales Group) to an extension in Extension
Configuration, and that feature profile is subsequently deleted and a new feature profile is created that uses the same number (for example: 2 – Marketing Group), the extension will automatically be assigned to the new feature profile. So, it is important to note which extensions are assigned to certain feature profiles, especially when adding new profiles or deleting old ones.
Limitation
You should include #26 (Station Logout) in a feature profile assigned to an AltiGen IP phone. If #26 is disabled in that phone’s feature profile, phone registration issues arise.
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Voice Mail Configuration
Use the Voice Mail Configuration window to control the following:
•
How the system processes voice mail notification
•
How the system processes voice mail deletion and expired messages
•
How the system records voice mail, system phrases, custom phrases, personal greetings, directory name recording, and queue phrases
•
Enable or disable SMTP/POP3 service to deliver voice mail to an e-mail address as an attachment
•
Enable or disable Microsoft Exchange synchronization service, or select Exchange’s bridged access or native VM integration with Exchange’s Automated Attendant or Unified Messaging Server
Note: A built-in throttle checks the amount of available disk space before allowing users to record a call. If the available space on the hard disk is less than 10% of the total space, the system will warn the caller that there is not enough room to record the call, and the call will not be recorded. This default threshold, 90%, can be adjusted (between 50% to 95%) by modifying a registry entry:
HKLM\SOFTWARE\Wow6432Node\AltiGen Communications, Inc.\AltiWare\DiskSpace; the default value is DWORD 90.
To open the Voice Mail Configuration window, select System > Voice Mail Configuration
Managing Messages
The Messaging tab in the Voice Mail Configuration window provides for setting basic parameters and options for messaging, including message notification retry attempts, message management options, recording options, and e-mail activation and usage.
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Figure 29. Voice Mail Configuration, Messaging tab
Setting Message Notification Retries
When a message is sent to a user’s voice mailbox and outcall notification is configured, the system will try to call a phone number, pager, or an extension to deliver notification. You can set the retry setting for the notification as follows:
Message
Notification
Parameter
Maximum Retry Count
Retry Interval in Minutes
Description
Can be between 0 and 16. This is the number of times the system will try to deliver a voice message notification after the original attempt. For example, 5 retries means five tries after the original, or 6 total attempts.
The number of minutes between retry attempts. Five minutes is the minimum and 60 minutes is the maximum interval allowed. Choices are in 5-minute increments. The default is 5 minutes.
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Managing Messages
Setting Message Management Options
Set voice mail message confirmation and warning parameters:
Message
Management
Parameter
Confirm Message
Deletion
Warn Expiration of Saved
Messages
Description
If checked, the system plays a voice message instructing the user to confirm request for deletion by pressing the # key. This prevents users from accidentally deleting messages with a single key entry.
If checked, the system warns the user that saved messages will be deleted due to their retention time expiring. The message is given the day before the messages are automatically deleted, and the user then has the option to either keep or delete the messages. By default, this feature is enabled.
Note: If this feature is disabled, saved messages are deleted automatically without warning when they expire.
Setting Message Recording Options
Set voice mail message recording parameters:
Message Recording
Parameter
Minimum Recording Length
Pause Detect Length
Description
Sets the minimum length in seconds for any recording (incoming voice mail message, personal greeting, system prompts, introductions to forwarded voice mails). This can be from 1 - 5 seconds, or 0, which means no minimum.
All recordings that are shorter than the designated Minimum Recording length are deleted. This feature is recommended when users receive many short, empty voice mail messages on a regular basis and would like them automatically deleted.
Selected, this feature causes the deletion of pauses in messages. The default pause detect length is 500 ms. The pause detect can be disabled by deselecting the check box, or the length can be set to a value between 200–
2000 ms (.2–2 seconds).
Setting Exchange Integration Options
Set Exchange integration options. Access to these options requires an AltiGen Exchange Integration License.
To assign this license to an extension, see “Assign Exchange Integration License” on page 192.
If you are opting to use Exchange’s Speech Enabled Voice Mail features or Unified Messaging, Exchange
Server and MAXCS need to be installed on the same domain with a network throughput rate of no less than
100 Mbps.
Full configuration details are given in “Microsoft Exchange Integration” on page 369.
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You may choose an option when you install MAXCS, and you may change the option later. If you change the option later, you need to restart services.
Exchange Integration
Parameter
Disabled
Synchronize with Exchange
Bridged Access to Exchange
Enable Synchronization
Native VM Integration with
Exchange
Exchange Client Access
Server
Exchange Unified Messaging
Server
Description
Disables Exchange integration.
Allows a two-way synchronization between a user's MAXCS voice mail and the user’s Outlook-readable mail messages with their attached .wav files in the user's Inbox. When e-mails or voice mails are deleted from one server, they are automatically deleted from the other server as well. (This is what
AltiGen’s previous releases have offered.)
If you select this option, enter the DNS name of the Exchange server in the
Exchange Client Access Server field (do not enter the IP address).
Through bridged access integration, via a SIP connection, AltiGen’s Voice
Mail System provides an option to the user to directly access Exchange Unified Messaging (by pressing 7 in extension voice mail after entering the password). Once connected, users can check and reply to e-mail, manage calendars, and send messages.
Note: If Exchange server is offline or down, the user pressing 7 will hear an error message, “This voice mailbox is not accepting new messages at this time.” All other VM features still function, as they are provided by the MAX-
CS system and not Exchange server.
If you select this option, enter the DNS name of the Exchange server in the
Exchange Unified Messaging Server field (do not enter the IP address).
If you want to synchronize voice mail between the AltiGen mail box and the
Exchange server, check the Enable Synchronization check box. If you don’t check this, voice mail is not synchronized between the two message stores.
Uses Exchange as a native voice mail box to store voicemail files, providing a unified mailbox for all message types. Callers are forwarded to the
Exchange mailbox when an extension is ring-no-answer, busy, or in DND.
Accessing voice mail is done through the Exchange system.
When this option is activated, all physical/virtual/WG mail boxes with associated Exchange mailboxes are switched to Exchange. Extensions that do not have an Exchange mail box are treated as mailbox disabled.
Users with an Exchange account press ## to log into the Exchange voicemail box. The system establishes a voice stream to the Exchange mailbox through a SIP connection.
To turn on the message waiting light on the desktop phone and allow Alti-
Gen CTI client applications to manage voice mails, the voicemail files are replicated back to MAX Communication Server. When a voicemail file is heard, marked save, or deleted from an AltiGen client application, the voicemail attribute is changed in the Exchange server accordingly.
Limitations:
•
Personal options usually invoked by pressing 4 on the AltiGen Voice Mail
System menu must be invoked by pressing #14.
•
The following AltiGen voice mail functions are not supported: activity greeting, voice mail distribution list, voice mail out call.
•
One Number Access is not available.
If you select this option, enter the DNS name of the Exchange server in the
Exchange Unified Messaging Server field (do not enter the IP address).
Enter the DNS name of the Exchange Client Access Server.
Enter the DNS name of the Exchange Unified Messaging Server.
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Creating Distribution Lists
Setting E-mail Messaging Options
To use the MAXCS e-mail services, configure the following settings.
E-mail Messaging
Parameter
Enable SMTP/POP3 E-Mail
Service
Postmaster Ext
Description
Selected, this enables incoming and outgoing mail services on MAXCS
— Simple Mail Transfer Protocol (SMTP) and Post Office Protocol
(POP3).
This field defines the extension that will be assigned as a Postmaster Extension. When the e-mail system receives an e-mail with an invalid e-mail account, the automatic reply to the sender (informing of the invalid e-mail account used) is sent from the defined extension.
Note: The system always requires an extension to be specified as the
Postmaster Extension. By default, the first extension in the system is used. If an extension is selected as the Postmaster
Extension, it cannot be deleted until the Postmaster Extension is re-assigned to another extension.
Creating Distribution Lists
The System Distribution Lists provide for forwarding voice mail messages to multiple recipients defined as list members. To forward a voice mail to all list members, a user needs to enter only the two-digit ID instead of entering numerous individual extensions.
You can create up to 100 distribution lists, each composed of up to 64 extensions. The extension list member can represent another distribution list.
Note: The system distribution lists discussed here are different from the extension distribution lists, which are configured through the phone sets or the MaxCommunicator or MaxAgent user applications.
To configure distribution lists, select System > Voice Mail Configuration, then click the Voice Mail
Distribution List tab.
Figure 30. Voice Mail Configuration, Voice Mail Distribution List tab
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Defining a Distribution List
1.
On the Voice Mail Distribution List tab, select an ID (00 – 99) in the System Distribution List ID list.
The list name, if any, now appears in the Name box; the members of the list are now displayed in the
Member box, and other available extensions are displayed in the Non-Member box.
2.
To give the list a name or change the existing name, type a descriptive name into the Name box.
3.
To add a member, select the name(s) in the Non-Member list and click the Add button to move it to the
Member list.
To remove a member, select the name(s) in the Member list and click the Remove button to move it to the Non-Member list.
You can select multiple names by using Shift-click or Ctrl-click.
4.
Click Apply to save your changes, or click OK to save and close the Voice Mail Configuration window.
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Auto Attendant Configuration
The auto attendant (AA) feature provides quick and courteous processing of all incoming calls. An AA can be configured to serve as a primary attendant or as a backup to a receptionist. In a call-heavy environment the
AA can greatly reduce the number of calls that need to be handled by the operator.
You can set up to 255 different AAs. AA features include:
•
Multiple levels of tree structure.
•
Repeat current level or jump to a specific level.
•
Transfer call to extension, workgroup, hunt group, or operator.
•
Dial by Name – allows a caller who does not know the extension number to spell the name using the telephone key pad. The system will search the Directory and make a match on the name to connect the caller to the intended party's extension.
•
Name Directory Service – allows callers to hear a list of employees and their extension numbers.
•
Records a voice mail message to a specific mail box.
•
Allows employees to call into the system and access voice from an external location.
•
Collects caller input data, for example, account code, ID, and so on.
•
Data-Directed Routing – Allows the routing of calls directed by the caller's input (digit or text).
•
Sets call priority and skill level requirement for workgroup call processing.
•
Other advanced features include System Call Back and routing calls to SDK-based add-on applications.
Planning Is Essential
Follow the steps below before you set up an AA.
1.
Before you configure tasks for one or more AAs, plan the entire setup. Decide how many options you will provide at each menu and how many menu levels you will use. Based on the choices in each menu, write down the appropriate prompts or phrases to play at each menu level.
2.
Record phrases for each menu level or use the pre-recorded phrases that are available to you. See
“Phrase Management” on page 85 for more details on how to record custom phrases, use pre-recorded
phrases and use professionally recorded phrases.
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Example: AA Planning
Auto Attendant ID:
100, Phrase 10
Main Menu for XYZ Office
Digit Meaning Action
1
2
3
4
5
6
7
Reserved for
Extensions
(no prompts)
Collect
Extension
Collect
Extension
Collect
Express
Support
Sales
Extension
Expand Tree
(No. 110)
Expand Tree
Technical
(No. 120)
Expand Tree
Support (No. 130)
Phone FAQs Expand Tree
(No. 140)
8
9
0
Operator To Operator
Planning is essential in organizing an AA menu structure that makes sense. Planning also helps you to identify needs for custom prompts.
This simple example, using sample work forms for each menu, shows a beginning structure: a main menu and two of the four expansions.
When callers are routed to workgroup extensions, the workgroups have their own call handling settings for greetings, update phrases, rules for sending to voice mail, and so on.
Timeout (not shown on forms): after
7 seconds on first level, call the operator; on any other level, go to top level by default.
*
#
9
0
7
8
5
6
Auto Attendant ID:
110, Phrase 20
Express Support
Digit Meaning Action
1
2
3
4
Installation Call Extension
(Workgroup 350)
Board Support Call Extension
Version 5
Support
Version 6
Support
(Workgroup 360)
Call Extension
(Workgroup 370)
Call Extension
(Workgroup 380)
Operator
Repeat Menu Repeat Level
Main Menu
To Operator
GoTo Top Level
*
#
9
0
7
8
5
6
Auto Attendant ID:
120, Phrase 30
Digit
Sales
Meaning Action
1
2
Hardware Call Extension
(Workgroup 310)
Applications Call Extension
(Workgroup 320)
3
Check Order
Status
GoTo Item 127
(Collect Order #)
4
Other:
Questions, etc.
Call Extension
(Workgroup 311)
Repeat Menu Repeat Level
Main Menu GoTo Top Level
Adding Auto Attendants
Attendants” on page 81. You don’t need to add a new AA if you’re going to use 16 or fewer.
To add an AA beyond the first 16,
Click the AA Configuration button, or select System > AA Configuration.
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Configuring Auto Attendants
Copies an AA to a selected
ID.
Exports all your AA settings to an HTML file
Figure 31. The AA Select window
•
Edit – Opens the AA window, where you can edit the selected AA.
•
Add – Opens the Add AA dialog box.
Select an ID in the list and type in a descriptive Name for the AA, then click OK.
•
Clear – Clears all edits to the selected AA, restoring system defaults.
•
Copy From – Lets you make a copy of an AA (and then modify it, as you like).
1.
Select your target ID from the AA Select window.
2.
Click the Copy From button.
3.
From the list, choose the AA you want to copy to your selected ID.
4.
In the pop-up box, click Yes to complete the copy.
•
Close – Closes the AA Select dialog box.
•
Help – Opens the help file for AA.
•
Export – Exports all AA settings to an HTML file.
Configuring Auto Attendants
To configure an AA, click the AA Configuration button, or select System > AA Configuration. When the AA
Select window opens, select an AA in the list and click the Edit button.
This opens the AA window, showing the AA you selected in the title bar.
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Figure 32. The AA window
Note: You can check the Hide ‘No Action’ Items check box to hide items that are set to “no action.” This will give you a cleaner view of your various action items.
Configuring Menu Items
The AA is a tree-based structure with unlimited tree levels. The following rules guide the basic AA configuration:
•
Each item is an action point with its ID number and name.
•
The top of the tree is a "O" (for Origin).
•
A timeout is indicated by a "T".
•
Any action item can have a "Prompt". The list displays phrase files located at
C:\Postoffice\Phrases\LangCustom directory. A phrase file can be any file name. (Note: Prior to the 5.1
Release, the "Phrase" directory was under C:\AltiServ, and custom phrases had to use a phrase number from 0001 to 0999.)
•
If one action item has multiple choices, you need to select "Expand Tree" instead of using "Go to next menu" to create a new level.
•
You can jump to any action item within the same AA.
Every item will execute steps according to the following rules:
•
First step – Play prompt if the box is checked. If the prompt box is not checked, the AA will go to the second step without delay.
•
Second step – Push URL/Web-Page forces a web page to a Web caller's screen when the call reaches the AA using AtliWeb's Webcall button over the Internet. If this box is not checked, go to the next step without delay.
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Configuring Auto Attendants
•
Third step – Set Call Priority for MAXCS ACM priority queuing. You can assign a priority number from 1-
9 to the caller who selects this menu item. The highest priority is 1, the lowest priority is 9. If this box is not checked, go to the next step without delay.
•
Fourth step – Set Call SKLR (Skill Level Requirement) for MAXCS ACM skill-based routing. You can assign an SKLR from 1-9 to the caller who selects this menu item. If this box is not checked, go to the next step without delay.
•
Fifth step – Execute the action selected from the drop down list. The following table describes each action.
Auto Attendant Actions
Action
No Action
Level – Expand Tree
Level – Repeat Current
Level
Level – Go to Top Level
Level – Go to
Specified Item
Call – To Ext./Group
Call – To Operator
Call – Dial By Name
Call – Collect Extension
Call – Directory Service
Call – Disconnect
VM – Record Message
VM – Mailbox Access
Adv. – System Call Back
Adv. – Collect Digits
Adv. – Advanced Call
Router
Adv. – Application
Process Control
Description
An “invalid” message plays and the menu is repeated.
Expand menu item to create additional level.
Repeats the level that contains the “Repeat Current Level” menu item.
Go to the top level and repeat action items on the top level.
Goes to selected menu item at any level. A list opens from which you select the item.
Transfers call to an extension or group number you select in the list.
Routes the call to the operator (the operator is defined in the System
Configuration window).
Prompts the caller to enter the name (first, or last, or both in any order) of the person they want to speak with and dials the extension that matches the name.
Callers may not have to enter the entire first or last name before a match is found.
The top level of each AA collects the extension number automatically. The system has a timing delay to differentiate if the first digit the caller entered is a menu option or the first digit of an extension number. Once past the top level, the system will not have the timing delay to differentiate digits. If you would like to provide the option for a caller to enter an extension number, you need to map this action item to one of the menu options.
Lists the system users and their extensions to the caller. For this to work properly, users need to record their directory names.
Disconnects the call.
Leaves a voice mail message in the specified voice mail box. If you want the caller to hear the extension’s greeting before hearing the start-recording beep, check Play Extension Greeting.
Allows the caller to log in to the voice mail system to retrieve voice mail or change personal options from the outside. This option is assigned to the "#" key at the top level of each AA by default.
Allows outside caller to dial into the system, enter a call back number, hang up, and wait for the system to call back. The system will request the caller to enter an extension and password for authentication. The call back number needs to include the toll call prefix and area code for long distance and international calls.
The trunk or route access code is not required when entering a call back number.
See the discussion on “Collecting Digits.
When selected, the system will hand over the call to the Advanced Call Router application through the SDK API interface. The ACR application needs to log in to a virtual extension with the correct password. If the ACR application fails to connect, the system will execute the sub-level "&" as a fail action.
When selected, the system will hand over the call to the APC (Application
Process Control) SDK through an application extension as a control extension.
An SDK APC based application needs to log in to the application extension to receive the call. If the APC application fails to connect, the system will execute the sub-level "&" as a fail action.
Note: The APC SDK license is not supported in the MAXCS Private Cloud service.
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Collecting Digits
When a caller selects the "Collect Digits" action item, a custom phrase is required to advise the caller how many digits are required. The system will look at "Min Length" and "Max Length" to determine if the collect digit action was successful or failed.
•
If successful, the system executes the sub-level "&" action item.
•
If failed, the system executes the menu item you define as a fail over action.
To use the Collect Digits action, select the Adv. – Collect Digits action, then set additional parameters.
Figure 33. Collect digits
•
Text Tag – A tag name, which is critical for the following operations:
•
For CDR logging, the IVRData field will log the collected digits as "Tag=xxxxx". For example, if tag is configured as "Account" and collected digits is "67663", the CDR database will log
"
Account=67663
" in the IVRData field.
•
For MaxAgent client display, the above example is displayed as "Account=67663" on the View >
IVRData section.
•
For CDR Search, the above example is displayed as "Account=67663" on the IVRData column.
•
To display collected digits on the AltiGen IP phone, you need to set the tag as "DISP" (stands for
"Display" and is case-sensitive. The Phone Display/Name Line of the extension configuration needs to be configured as IVR Data (Display). This feature supports inbound trunk calls only.
•
Min. Length – The minimum length of digits to be collected.
•
Max. Length – The maximum length of digits to be collected.
•
PSTN Call Inter-Digit Timeout – The length of time the system will wait between collecting of digits before timing out.
•
Inter-Digit Timeout after Max Length – The length of time the system will wait after the maximum length of digits is collected.
•
Web Call Response Timeout – The length of time the system will wait for digits after responding to a
Web call before timeout.
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Phrase Management
Making Auto Attendant Assignments
Once the AAs are set up, you can use them in various in-call routing situations – trunk, DNIS, caller ID, in-call routing, and an answering option for an extension or workgroup.
assignments, see “Setting Answering Options” on page 199.
Phrase Management
You might want to record unique phrases to customize an AA or a group. When the system is configured to have the AA answer incoming calls, callers hear a customized greeting. For example:
“Thank you for calling ABC Company.
Enter the extension number of the person you wish to speak with.
Press 1 for sales.
Press 2 for technical support.
Press 3 for accounts payable.
Press 0 to reach the operator.
To repeat this menu, press star (*).”
An example of a group greeting phrase:
“Please hold; someone will be with you shortly.”
You might also want to give callers the option of hearing prompts in another language. For information on
configuring for a multilingual AA, see “Multilingual Configuration” on page 89.
This section covers information on how to use pre-recorded phrases, record custom phrases, and use professionally recorded phrases.
Using Pre-Recorded Prompts
MAXCS provides ready-to-use pre-recorded phrases. Phrase 0001 is the default AA greeting at the root menu level. Phrases 0291 through 0297 are phrases used for group queue prompts. Select the phrase you want to use in the Prompt field. To hear the pre-recorded phrases:
1.
Use any phone to dial “###”, and log in with the system manager’s extension and password.
2.
Press 6 for the Phrase Management option.
3.
Press 1 to review a phrase.
4.
Enter the 4-digit phrase number from the list below to hear the phrase.
Phrase #
0001
(default)
0291
(default)
0292
0293
0294
0295
0296
Phrase
Thank you for calling. If you know the extension of the person you wish to speak with, please enter it now. To reach the operator, press 0 or simply stay on the line.
Please hold; someone will be with you shortly. For your convenience, you may leave a message if you wish by pressing the # key on your telephone and we will get right back to you.
Please hold; someone will be with you shortly.
We appreciate your call and will be with you as quickly as possible.
Thank you for your patience. We should be with you soon.
Thank you for your patience. We should be with you soon. For your convenience, you may leave a message if you wish by pressing the # key on your telephone and we will get right back to you.
We apologize for the extended delay, but our current call load is abnormally high. Remember, you may leave a message by pressing the # key on your telephone and we will get right back to you.
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Phrase #
0297
0400
0401 callback callback1
Phrase
You may still wait if you prefer, but we suggest you leave a message by pressing the # key on your telephone and we will get right back to you.
Please hold. Someone will be with you shortly.
Thank you for holding. Someone will be with you as soon as possible.
You can also press 1 to schedule a callback.
This is a callback for…
Recording Custom Phrases from the AltiGen Phone
Note: If you have an AltiGen SDK license, you can use the AltiGen Custom Phrase Manager discussed in
“AltiGen Custom Phrase Manager” on page 419. This application has a graphical user interface that
makes recording phrases easier.
When you create custom phrases from the AltiGen phone, keep a record of phrase numbers and the corresponding phrases so that if a phrase needs to be changed, the correct phrase number is readily available.
To record a custom phrase,
1.
Log in from any phone on the system by dialing “###”, and entering the system manager’s extension and password. This brings you to the AltiGen Voice Mail System Main Menu.
2.
Press 6 for the Phrase Management option.
3.
Press 2 to record a phrase.
4.
Enter a four-digit phrase number between 0001 and 0999.
5.
Record the phrase after the tone. Press # at the end of the recording.
6.
The system will replay the recorded phrase. Press # if the recording is acceptable.
7.
At the Phrase Management menu, press 2 to record additional prompts or star (*) to exit Phrase
Management.
Phrases are stored in the C:\PostOffice\Phrases\LangCustom directory. You can modify the phrase file to any meaningful name if you want.
Using Professionally Recorded Phrases
Recording studios such as Worldly Voices provide professionally recorded prompts as electronic files that can be installed and used on the MAXCS system. (See the AltiGen web site, at www.altigen.com, for more information. Click Customer at the top of the page, and then click Resources for Creating Professional
Voice Prompts.)
AltiGen provides the Voice File Converter utility to convert these files into the proper MAXCS format (use the
Windows Start > Programs > MAX Communication Server ACM > Utilities menu). Some recording studios provide the conversion service for an additional fee. The converted file can then be used for an AA or for a workgroup or huntgroup group setup.
To install professionally recorded phrases or prompts,
1.
Assign a prompt number to each prompt you would like recorded. Or give the prompt a unique identifying name. AltiGen-supplied phrases are numbered, but phrases don’t have to be numbered.
2.
Submit your prompt script and prompt name to the recording studio.
3.
Instruct the recording studio to record prompts in either 8KHz or 11.025KHz mono in the WAV format.
4.
Ask the studio to convert the WAV file(s) into the proper MAXCS format.
•
If using Worldly Voices, this conversion is done for you.
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Phrase Management
•
If you are using a studio other than Worldly Voices, use the Voice File Conversion utility. This utility converts an audio file recorded at either 8KHz or 11.025KHz in the WAV format to an MAXCS playable audio file.
5.
Once you receive the prompts in the MAXCS format, place them in the
C:\PostOffice\phrases\LangCustom directory on the gateway that is running MAXCS.
Your prompts are now ready to be used.
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7
Multilingual Configuration
MAXCS supports multiple language prompts (8 languages maximum) for trunk calls and extension users, letting you configure your system to handle the following types of scenarios in a multilingual environment:
•
An auto attendant (AA) may serve callers who speak different languages. MAXCS can be configured to let the caller select a preferred language in which to hear prompts. Once a language is selected, the whole call session will use the selected language.
•
An internal user may use a feature code to execute a certain action, including logging into voice mail.
Normally the user hears system prompts first. If the user is not fluent in the default system language, another language can be assigned to his extension. Whenever that extension user encounters prompts, the system will use the assigned language to play the prompts.
•
DNIS may also be used to select a language for the caller. If your company has multiple phone numbers, you can configure MAXCS to direct a caller to a language based on the phone number the caller has dialed. For example, if you give out different 800 numbers to different countries, and a call comes in from the 800 number you give out to customers in Mexico, you can configure MAXCS to direct that 800 number to the “Mexico Spanish” language prompts or to an extension that uses the corresponding language in its prompts. This eliminates the caller having to select a language.
Note: The MAXCS multilingual feature requires the purchase of an AltiGen Multilingual License.
Configuration Overview
Configuring multilingual features involves most or all of the following actions, which are discussed in subsequent sections:
•
Have the appropriate system and custom phrases recorded in each language that your company wants to use (in addition to the default language).
•
Store the custom phrases in new directories under the C:\PostOffice\Phrases directory, using the prescribed naming convention.
•
Add the new languages to the Multilingual Configuration screen.
•
Enable auto attendant support in the Multilingual Configuration screen, AA tab.
•
In the Extension Configuration screen, choose an available language for the internal user, if desired.
•
Enable the extension user to change the preferred language for the extension by using a feature code
#12, if desired.
•
Configure the Language Setting in DNIS, if desired.
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Creating Language Phrase Packages
For each set of phrases you want in a different language, you need to have phrases recorded in that language.
contain phrase files, and two text files: one text file that lists syntax rules for numbers, and one that lists syntax rules for sentence structure, since these vary from language to language.
The phrase files will have the exact same name/number as in the default language directory and will be part of the same AA, but they will be stored in a different directory.
Note: AltiGen authorized distributors in each country will perform localization procedures to create language packages, including syntax rules for numbers and sentence structure for their local market. For international customers, please contact the authorized AltiGen distributor in your country to obtain localized language phrases.
Storing Language Phrase Packages
Additional language phrases (system and custom) and syntax styles need to be copied to the correct directory before system startup, so that the system can recognize them. If they are added after system startup, MAXCS needs to be shut down and restarted, before the directories are recognized.
(see Figure 34) illustrates the directory storage structure for language phrases.
Figure 34. Storage structure for multilingual phrases
The directories
Lang1
and
LangCustom
contain the phrases of the system default language.
Phrases for language X should be saved in a pair of directories: Lang_X and LangCustom_X
.
Lang_X stores the phrases required by the system, and LangCustom_X stores your custom phrases.
For example, to add a language for Mexico, you need to create two directories:
•
Lang_Mexico
•
LangCustom_Mexico
Configuring for a Multilingual System
To configure MAXCS as a multilingual system, select System > Multilingual Configuration. The configuration screen opens to the Language tab. Here, you will add references to the language directories you created. These are the directories that contain phrases in other languages.
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Enabling Multilingual Support in the Auto Attendant
Figure 35. Multilingual Configuration, Language tab
When you first run MAXCS , only the default language is listed in the Multilingual Configuration screen, and the description of the default language is displayed as Default Language. Each language added to the table will have a formal name, a description, a system phrase directory (LangDir_X), and a custom phrase directory
(LangCustomDir_X), as shown in (see Figure 35).
To add a language,
1.
Click the Add button.
2.
Choose a language from the list. The list shows the language directories you have added to the
C:\PostOffice\Phrases directory.
3.
Enter a description for the language. This description will appear elsewhere in the graphical user interface, for example in the Extension Configuration window and the AA tab in this screen. Click OK.
4.
Repeat these steps for each language you want to add.
The contents of the fields System phrase directory and Custom phrase directory are fetched from the location where the language phrases are stored. They are not editable.
Only the description of the language is editable here. To edit it, click the Edit button or double-click the row.
The default language cannot be deleted. After you add languages, any language used by DNIS, an extension, or an AA cannot be deleted.
Enabling Multilingual Support in the Auto Attendant
After you have recorded phrases and added a reference to their directories in the Multilingual Configuration
> Language tab, as described above, you are ready to enable multilingual support in the auto attendant.
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1.
Select System > Multilingual Configuration > AA tab.
2.
From the list at the left, select the AA you want to configure with multilingual support.
3.
Check the Enable Multilingual Support check box. The Multilingual Enabled column changes to
TRUE.
4.
In the Language Setting group of fields, check the Language Selection Prompt check box.
5.
Choose the prompt that lets the caller select a language.
6.
Beside each appropriate number, select a language from the list that corresponds to the phone key the user would press to hear that language. (For example, “For English, press 1; for Spanish, press 2...”)
7.
Click Apply if you have more work to do in the configuration screen, or click OK to accept the changes and close the screen.
Note: This configuration is on top of the regular AA configuration. The system will execute the regular AA action items after a language preference is selected by the caller.
Configuring the Extension
Extension users have a default language configured, and that language is always used for them whenever they hear a prompt on their extension. The default language is assigned in Extension Configuration > General tab.
Figure 36. Selecting a language for an extension user
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Configuring the Extension
In the Language list, select the desired language, and click OK.
Extension User Can Change Language Setting
Extension users can change the extension’s language setting by using feature code #12, if feature code #12 is configured on the System > Multilingual Configuration > Feature Code tab.
Figure 37. Configuring feature code #12 to allow a user to change an extension’s selection
Configuring Feature Code #12 for Language Selection
1.
Check the Language Selection Prompt check box.
2.
Select the prompt the extension user will hear after pressing #12. You must know the text of this prompt, so you can match the languages to the correct numbers in the next step.
For example, the prompt the extension user might hear after pressing #12 might be “To change the preferred language for this extension, press 1 for English, press 2 for Spanish, press 3 for Chinese.”
3.
Beside each number, select a language from the list that corresponds to the prompt. The languages listed are those that you have added to MAXCS on the Language tab of this window.
For example, if you were working from the example prompt in step 2, you would select English beside the number 1, Spanish beside the number 2, and Chinese beside the number 3. The remaining fields would be left as None.
Feature code #12 must also be enabled in System Configuration > Feature Profiles tab.
To enable feature code #12,
1.
In System > System Configuration > Feature Profiles tab, check the #12 -language setting check box.
2.
Click OK.
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Allows extension user to specify his extension’s language setting using #12
All feature codes are enabled, by default. The default feature profile name is “0-System.”
Lastly, the extension user must have a feature profile assigned to him that includes #12. This is done on the
Extension Configuration > General tab.
To assign feature code #12 to an extension,
1.
On the PBX > Extension Configuration > General tab, select the extension.
2.
In the “Personal Information” panel of the General tab, assign a Feature Profile that includes #12.
Using DNIS to Set the Language
If your company has multiple phone numbers, you can configure MAXCS to direct a caller to prompts in a selected language based on the phone number the caller has dialed.
To direct specified DNIS calls to a selected-language AA or extension,
1.
Select PBX > In Call Routing Configuration > DNIS Routing tab .
2.
Click the Add button to add a number.
3.
Select where you want to route callers who have dialed that number.
4.
Select the appropriate language from the Language Setting list.
5.
Click Apply.
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Which Language Will Be Used?
Figure 38. Configuring the language setting in DNIS
See “DNIS Routing” on page 165 for rules and restrictions on routing using DNIS.
Which Language Will Be Used?
MAXCS follows these rules to determine which language to use:
1.
The extension user hears the prompts in the language configured or selected via the #12 feature code.
2.
If the external caller selects a language in the auto attendant, MAXCS uses the selected language. If a language selection is invalid or times out (7 seconds) three times in a row, the default language is selected.
3.
If an extension is set for ONA (one number access), the caller will hear the prompt in the language selected previously, but when the callee picks up the ONA notification call, the callee will hear the prompt in the language according to the extension's language setting.
4.
When the user logs in to the voice mail of an extension, the extension's language is used.
5.
If DNIS is configured for language setting, the external caller hears the prompts in the language specified by the number he dialed.
6.
In any other case, the system default language is used.
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Call Recording Configuration
To use the centralized call recording function, make sure the following requirements are met:
•
You need a recording seat license for each extension that will be recording: either Dedicated Recording
Seat licenses assigned to particular extensions or a Concurrent Recording Session license that is shared by a fixed number of extensions.
•
It is recommended that you have a separate storage server to store recorded files.
•
Recorded files (64 Kbps PCM format) can be managed by the VRManager (licensed) application or can simply be saved and played with VRPlayer (free).
•
If your system has a multi-chassis configuration and the gateway needs to transmit recorded files to a storage server, you need to set up an FTP server to facilitate the file transfer. You do not need to set up an FTP server for a single chassis (all-in-one) installation.
•
If an agent is using an IP phone and recording is turned on, the system will use a recording channel on a
VoIP board to process the recording session. The IP phone will occupy a codec channel on the VoIP board to allow the recording channel to tap into the conversation. You need to make sure that the MaxCS 7.5 that agents belong to (and the gateway for a multi-chassis installation) have adequate VoIP codec channels to record conversations. The basic guideline is to have one codec channel per agent.
•
Because recording files requires a large amount of disk storage space, NAS (Network Attached Storage) system is recommended, unless VRManager is used.
Note: A built-in throttle checks the amount of available disk space before allowing users to record a call. If the available space on the hard disk is less than 10% of the total space, the system will warn the caller that there is not enough room to record the call, and the call will not be recorded. This default threshold, 90%, can be adjusted (between 50% to 95%) by modifying a registry entry:
HKLM\SOFTWARE\Wow6432Node\AltiGen Communications, Inc.\AltiWare\DiskSpace; the default value is DWORD 90.
Polycom VVX models can also record calls; you can enable a Record softkey for those models. This option is found on the PBX > Extension Configuration Polycom tab. Once this feature has been enabled, the extension user can tap a Record softkey on their Polycom phone record the call in progress. This feature requires a Polycom Advanced Features license; see the MaxCS 8.0 Polycom Configuration Guide for details.
Description of the Recorded File Name
The recorded file name contains the following information:
•
R!mmddyyyy_hhmmss!callerID!calleeID!workgroupID!DNIS!sessionID!R
•
mmddyyyy_hhmmss is the time stamp when the recording starts
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Chapter 8: Call Recording Configuration
•
callerID is the caller ID or extension number. It could also be:
•
bgn for barge-in call
•
sm for a silent monitor call
•
trk(bbcc) for an inbound trunk call without caller ID. bb is the board logical ID and cc is the channel ID
•
calleeID is the target number or trk(bbcc)
•
workgroupID is the workgroup number for a workgroup call, or ext for extension call
•
DNIS is the DNIS number or NA for no DNIS number
•
sessionID is the CDR session ID
Configuring Call Recording
Note: During recording, any communication from a barge-in or coach supervisor's extension becomes part of that call recording.
To configure system-wide call recording, including centralized recording for multiple gateways, do one of the following:
•
Click the Recording button on the toolbar.
•
Select System > Call Recording Configuration.
The Recording Configuration window opens:
Figure 39. Recording Configuration window
Note: Call recording options for specific extensions/workgroups can be set up on the General tab of
Extension Configuration and Workgroup Configuration, respectively.
To enable centralized recording,
1.
Check the Enable Centralized Recording check box.
2.
Select a Recording Type from the drop-down list.
3.
In the Central Location field, browse for the directory you want to set as the destination folder and path for saving the call recordings.
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Important: If you are using FTP protocol, the FTP server must be installed and configured properly on the same machine as the Central Location directory. An FTP folder must be created for the Central
Location, so that it can be fully accessible through FTP. The FTP Path must be pointed to the
Central Location.
4.
If you are using multiple gateways, and you are not using network attached storage, check Gateways Use
FTP Protocol to Transmit Recorded Files to Central Location.
a.
FTP Server – Enter the IP address of the FTP server.
b.
FTP Access Account – An FTP server account name that gateways can log in to.
c.
FTP Path – Enter the directory that the files will be transmitted to on the FTP server.
d.
Password – FTP account password.
5.
Click the FTP Test button to verify that login to the FTP server is successful.
6.
When you are finished configuring, click OK.
Note: To allow supervisors to record an agent’s non-workgroup call, check the appropriate check box on the
System Configuration General tab. For more information, see “Setting General Parameters” on page
Using a Remote Shared Directory
It is strongly recommended that you use VRManager to manage centralized recording and that you save recordings to a local drive or network attached storage on the gateway that is running MaxCS 7.5 . If you save recordings to a network drive, and the network becomes unstable, you could lose any files of conversations being recorded at that time.
However, if you need to use a remote shared directory, and you are using Windows Server, follow the steps below:
1.
From the desktop, select Map Network Drive from the Tools menu.
Figure 40. Map Network Drive
2.
In the dialog box, click the Sign up for online storage or connect to a network server link. This starts the Add Network Place Wizard.
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Figure 41. “Sign up for online storage or connect to a network server” link
Figure 42. Add Network Place Wizard
3.
Click Next. You’ll see the screen below:
Figure 43. Add Network Place Wizard
4.
Click Choose another network location and click Next.
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Figure 44. Add Network Place Wizard – Internet/Network Address
5.
Type the address of the Web site, FTP site, or network location in the field, for example,”\\Server-
Name\sharefolder”; or use the Browse button to locate the destination path. Click View some examples for correct formatting. Then click Next.
Figure 45. Add Network Place Wizard – Shortcut Name
6.
Type in a name for the network place and click Next. A confirmation screen opens:
Figure 46. Confirmation screen
7.
Click Finish. The network place you created should appear on the desktop.
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Figure 47. Network Place Created
8.
In the Recording Configuration window, use the Browse button to select the network place as the destination folder.
Figure 48. Recording Configuration Window
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9
Application Extension Configuration
The application extension is an extension pilot number that allows an SDK-based add-on application to log into the system and establish a communication channel to control trunk channels and interact with the system core
PBX switching and voice processing service.
Typical applications that use an application extension are:
•
IVR
•
Outbound dialer
•
Inbound call routing logic for a special business application
To connect an SDK-based add-on application, you need:
•
An APC license (concurrent session)
•
A separate application extension to log in to for each application
For more information about SDK, please send e-mail to [email protected].
Note: The APC SDK license is not supported in the MAXCS Private Cloud service.
Application Extension Setup
Note: Before you begin, make sure a Trunk Control APC SDK Session license is registered and activated for your system. You can find this information in License > License Information.
To access the Application Extension Configuration window, select System > Application Ext
Configuration. The configuration window opens:
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Figure 49. Application Extension Configuration window
To set up an application extension,
1.
In the Application Extension Configuration window, click the Add button and enter an extension number in the dialog box. Click OK.
Figure 50. Add Application Extension dialog box
2.
The application extension appears in the AppExt List.
3.
Type a password in the Password field.
4.
Type a description of the application in the Description field, if desired.
5.
Click OK.
Application Failover Plan
The Application Failover Plan ensures that a call made to the extension will be automatically transferred if the application is not available. Use the If application is not available, forward to list to select the forwarding destination. The options are:
•
AA – Select the auto attendant number to use in the list under the option. AA settings are configured in
System > AA Configuration.
•
Extension – Select an extension from the list.
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•
Operator – Select an operator from the list.
•
Disconnect – Disconnect the call
Important:
If the failover setting for the application extension is set to an extension, and the extension is RNA or busy, the call will follow the extension's RNA or busy call handling.
Application Information
Additional information can be described in the App Information fields. If desired, enter the appropriate information in the fields for Application Source, Spec Doc Location, Designed by, Implemented by,
Implementation Date, Revision Number and Revision Date.
Readying the Application
If a third-party application is connecting to this extension, make sure the application is properly set to log into
changes to “connected.”
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10
Board Configuration
This chapter shows how to configure AltiGen telephony boards:
•
Triton Resource Board: see Using the Triton Resource Board
•
Triton 30-Party Conference Board: see “Using the Triton MeetMe Conference Board” on page 109
•
Triton Analog Station Board: see “Configuring the Triton Analog Station Board” on page 109
•
Triton Analog Trunk LS/GS and LS Boards: see “Configuring the Triton Analog Trunk LS/GS and LS
•
Triton VoIP Board: see “Configuring the Triton VoIP Board” on page 110
•
Triton T1/E1 Boards: see “Configuring the Triton T1/E1 Board” on page 111
•
Virtual Board SIP: see “Configuring Virtual Board SIPSP” on page 123
•
Virtual Board HMCP: see “Configuring Virtual Board HMCP” on page 126
•
MAX1000/2000 Board: see “Configuring the MAX1000/2000 Board” on page 132
•
Virtual MobileExtSP Board: see “Configuring the Virtual MobileExtSP Board” on page 134
For information on how to install AltiGen boards, refer to the Quick Installation Guide provided with every board package.
Configuring Boards
Board attributes and functions are accessible from the Boards window.
Figure 51. Boards window
Double-click the board you want to configure, and a Board Configuration window opens, similar to the following figure.
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Figure 52. Board Configuration window
Board Configuration Parameters
These are the attributes and buttons in the Board Configuration window (see each board type in the sections that follow for additional notes on each type):
Parameter
Board Info
Channel Mapping List
Channel Group Info
Board Configuration button
Reset Board button
Description
Board Logical ID: assigned by MAXCS.
Board Name: the type of board installed in the system and its physical
ID.
Logical Channel, Type, and Physical Channel for the entire board.
Double-click a channel to open a line configuration dialog box or a trunk configuration dialog box, as appropriate.
To reset the channel, select the channel to reset and click the Reset
Channel button, then click OK.
Applicable to T1/E1 and the MAX family of boards only.
Double-click a channel group to open a configuration dialog box.
To reset a channel group, select it and click the
Reset Channel Group button.
Opens a configuration dialog box.
Resets the board, after you confirm.
Important! Resetting a board will disconnect all calls in progress on that board. Be sure to inform all users before resetting a board. Additionally, if the board is a resource board (VoIP 12 port, VoIP 30 port, Triton resource board, 30-party conference board), resetting it will disconnect all calls that use the resource.
Important:
To implement some board configuration changes, you must shut down and restart by choosing
Services > Shut Down All Services (which also closes MAXCS) and then restarting MAXCS.
If this is necessary, a message will pop up telling you so.
Using the Triton Resource Board
The Triton resource board requires no configuration. Board resources are available when the board is installed.
The resource board has a maximum of 12 bridges for:
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•
6-party conferencing. When an extension is trying to make a conference call, the system will try to use the conference bridge on the resource board first. If conference bridges on the resource board are all busy, the system will use the conference bridges on the extension board, analog or VoIP board.
•
Workgroup supervisor silent monitoring, barge-in, and coaching.
For example, if two supervisors are coaching agents, only 10 bridges are left for 6-party conferencing.
Notes:
If a supervisor tries to perform silent monitoring, barge-in, or coaching and there is no resource board in the
system,
the supervisor will hear an error tone.
If the supervisor is using an IP phone, then Connect Voice Stream to Server should be checked in the
Extension Configuration window so that the system can pull the caller and agent's voice stream to the resource board to allow the supervisor to tap into the conversation.
This option is in the
Extension Configuration window
Using the Triton MeetMe Conference Board
The Triton MeetMe conference board requires no configuration. Board resources are available when it is installed. You do have to assign a MeetMe Conference extension (select PBX > MeetMe Conference Config-
uration).
One MeetMe conference board is supported in a system.
Configuring the Triton Analog Station Board
Double-click the Triton Analog Station board in the Boards window to open the Board Configuration window,
similar to Figure 52. Note the following additional information:
•
Double-clicking a channel in the Channel Mapping List opens a Triton Analog Line configuration dialog
box. See “Triton Analog Station Line Properties” on page 185.
•
Clicking the Board Configuration button opens a configuration dialog box that displays the board’s serial number, DSP clock, physical and logical IDs.
Figure 53. Board Configuration dialog box
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Configuring the Triton Analog Trunk LS/GS and LS Boards
The Triton Analog Trunk board is a long form factor PCI telephony card that supports 8 or 12 trunks. The 8 port card supports only loop start (LS). The 12 port card is available in two models; loop start/ground start (LS/GS) and LS. Both models have the same features regarding LS. The LS/GS board is required when ground start trunks may be required.
Double-click the board in the Boards window to open the Board Configuration window, Note the following additional information:
•
Double-clicking a channel in the Channel Mapping List opens a channel configuration dialog box. See
“Triton Analog Station Line Properties” on page 185.
•
Clicking the Board Configuration button opens the following dialog box that displays the board’s serial number, DSP clock, physical and logical IDs.
Figure 54. Board Configuration dialog box
Configuring the Triton VoIP Board
VoIP for MAXCS runs on SIP protocols that allow voice calls to be made through an IP network. It includes an integrated VoIP gateway to convert voice calls into IP packets and transmit them through the IP network.
MAXCS VoIP uses DSP engines residing on the Triton VoIP board to perform the voice coding/decoding functions needed for SIP devices.
The Triton VoIP board can be configured as a 12-port G.711/G.723.1/G.729AB or 30-port G.711 board.
For limitations on configuring Triton VoIP boards and ports see AltiGen’s Telephony Hardware Manual.
To configure the board,
Double-click the TritonIP board in the Boards window to open the Board Configuration window, similar to
(see Figure 52 on page 108). See the attribute descriptions in Board Configuration Parameters.
Note the following additional information:
•
Clicking the Board Configuration button opens a window that displays the board serial number, DSP clock, and physical and logical IDs. The list in the Configure Type field lets you select between a 12-port
G.711/G.723/G.729 configuration and a 30-port G.711 configuration.
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If you change this configuration, you must restart the switching services for the change to take effect.
Figure 55. Board Configuration window
Configuring the Triton T1/E1 Board
Through MaxAdministrator, the Triton T1/E1 board can be configured for either digital T1 CAS (channel associated signaling), T1 PRI (Primary Rate Interface), E1 CAS, or E1 PRI.
Both T1 CAS and T1 PRI carry 24 channels using time-division multiplexing (TDM) at an overall rate of 1.544
Mbps. Voice T1 provides 24 64K channels with robbed bit signaling. T1 PRI provides 23 64K channels, using one 64K channel for D channel messaging.
E1 CAS and E1 PRI carry 32 channels using TDM at an overall rate of 2.048 Mbps. Both of them provide 30
64K channels for voice.
To subscribe to T1 CAS, T1 PRI, E1 CAS, or E1 PRI service, you must supply certain parameters. These
parameters are listed in Appendix B on page 443.
Configuring the Board
Double-click the Triton T1/E1 board in the Boards window to open the Board Configuration window, similar
information:
•
The Board ID must be in the range 0 - 7.
•
Double-click a channel in the Channel Mapping List to open a trunk configuration dialog box.
•
Double-click a channel group to open a configuration window, discussed in the following section.
•
Clicking the Board Configuration button opens a configuration dialog box that displays the board’s serial number, DSP clock, physical and logical IDs.
You can configure the board type: either T1 or E1 to run T1 CAS, T1 PRI, or E1 CAS, E1 PRI. Additional steps are needed to further configure the CAS or PRI protocol in the Protocol Configuration window,
shown in Figure 59 and Figure 60.
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Figure 56. Triton T1/E1 Configuration dialog box
T1 and E1 Configuration
Double-clicking a channel group for a Triton T1 board in the Channel Group Info pane opens a T1 or E1
Configuration dialog box, as in (see Figure 57) and (see Figure 58).
Figure 57. Triton T1 configuration dialog box
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Figure 58. Triton E1 configuration dialog box
Reading the Status Messages
If the channel group is working, the Status line displays OK. This status line is updated every 3 seconds. If there is an error, a message is displayed. The following table lists the types of error messages and the appropriate actions.
Error Message
HW failure: <No Answer>
HW failure: <No Clocks>
L1 failure: <No Signal
(LOS)>
L1 failure: <Alarm Indication
Signal (AIS)>
L1 failure: <Remote Alarm
Indication (RAI)>
L1 failure: <No Sync Frames
Meaning
Major hardware problem. Board is not responding to commands.
Reasons could be: 1) DSP loading failure; 2) If PRI, board failed.
Action
1.
Reset board.
2.
If error continues, replace board.
No clock signal is detected on T1 interface drop.
Layer 1 failure, physical layer;
LOS = Loss of Analog Signal
Layer 1 failure, CO sends all 1’s to our T1/E1; AIS = Alarm Indicator
Signal; all ones detected
Layer 1 failure, CO notifies that the configuration is wrong; RAI =
Remote Alarm Indicator
1.
Check MVIP clock.
2.
Reset board. If this does not work, replace board.
Check T1/PRI cable and change if necessary. If cable is okay, CO is not sending any signal. Contact CO.
To locate the AIS alarm, have the carrier check the T1 network element connected to the T1 interface and trace the problem.
Correct the settings.
Layer 1 failure, physical layer; no valid framing is detected.
Possible span mis-configuration (ESF is selected but the actual framing is SF, or vice versa). Check span configuration.
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Error Message
L1 failure: <Red Alarm>
[PRI only]
L2 Failure: <No Sync Flag>
[PRI only]
L2 Failure: <Not established>
Layer 1 failure, physical layer; Bi-
Polar Violations (BPV), Line Code
Violations (LCV), or Out Of Frame detected
Meaning Action
Location condition, equipment problem.
- For excessive BPV/LCV, check AMI/
B8ZS setting.
- For OOF, check the MVIP bus master setting.
OR
Have CO perform a line test to check for a faulty cable or line.
Layer 2 failure, data link layer; no sync flag has been detected in data link layer
Layer 2 failure, data link layer; the peer-to-peer link has not established in data link layer
Check if D-channel is active or not
CO must activate HDLC link
Reading the Statistics
The Statistics panel displays the number of errors that have occurred since the last system reboot or statistics clearing. There may be non-zero values when configuring the T1 span for the first time. You can clear these fields with the Clear button.
Frame Errors
Error
OOF Errors
Rec Frame Slips
Line Code Errors
Bit Errors
Xmt Frame Slips
Clear button
Meaning
Number of framing bit errors. In T1 mode, a framing bit error is defined as an incorrect FS-bit value. The counter is suppressed when framer loses frame alignment
The Out Of Frame counter registers every time the T1 chip is forced to reframe when receiving a frame with severe errors.
The Receiver Frame Slips counter shows the number of frame slips for the receiver.
Line Code Error is defined as an occurrence of a bi-polar variation or excessive zeroes.
Bit Errors are defined as a CRC-6 error in ESF, FT-bit error in SLC-96 and
F-bit or sync bit error in SF.
Transmit Frame Slips counter shows the number of frame slips for the transmitter
Use the Clear button to reset the statistics counters.
Note: For ideally synchronized systems, Transmit and Receive Frame Slips counters should be ‘0.’
Continuous update of the frame slips counters means that transmit and receive frequencies are not equal. In this case, you should check the system and CT-Bus clock setup.
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Setting the Configurable Options
These are the options you can set:
Line Code
Option
Frame Type
Zero Code Suppression
CD Bits Handling
System Clock Master
Notes
•
For T1, you can set the Frame Type to either SF or ESF. SF
(Superframe Format) consists of 12 consecutive frames. ESF
(Extended Superframe Format) consists of 24 consecutive frames.
•
For E1, you can set the Frame Type to either No CRC or CRC4. CRC4 is embedded into 16 consecutive frames.
•
For T1, you can set the Line Code to either AMI or B8ZS. AMI (Alternate Mark Inversion) is the line coding format in T1 transmission systems whereby successive ones (marks) are alternately inverted and sent with opposite polarity of the preceding mark. B8ZS (Binary 8 Zero
Substitution) sends two violations of the bipolar line encoding technique, rather than inserting a one for every seven consecutive zeros.
•
For E1, you can set the Line Code to either AMI or HDB3. HDB3 (High
Density Bipolar Order) is based on AMI, but extends this by inserting violation codes whenever there is a run of four or more zeros.
You can set the Zero Code Suppression to None (default setting), Jam
Bit 8, GTE or Bell.
Zero Code Suppression inserts a “one” bit to prevent the transmission of eight or more consecutive “zero” bits; Jam Bit 8 forces every bit 8 to a one;
GTE Zero Code Suppression replaces bit 8 of an all zero channel byte to a one, except in signaling frames where bit 7 is forced to a one. Bell Zero
Code Suppression replaces bit 7 of an all zero channel byte with a one.
CD Bits Handling is not editable.
You can set the System Clock Master if you have a back-to-back configuration and you want this span to be the master clock to the system. (Only one clock master should be selected in a back-to-back system.) See the following section on T1/E1 clocking.
T1/E1 Clocking
Depending on the configuration of the T1/E1 boards and span for your MAXCS system(s), the System Clock
Master setup should be set according to the follow conditions:
•
If all of the T1/E1 boards are connected to a carrier’s switch, the System Clock Master check box must
not
be checked for any of the T1/E1 boards.
•
If two MAXCS systems are connected back-to-back with a T1/E1 span, the System Clock Master check box must be checked for only one of the T1/E1 boards.
•
If two T1/E1 boards in the same MAXCS system are connected back-to-back with a T1/E1 span, the
System Clock Master check box must be checked for the T1/E1 board that has not been designated by the CT-Bus setting as the system’s master clock to drive the CT-Bus.
Important:
For all back-to-back cases, the CT-Bus Clock Configuration should be set to “Manual,” and the board that is connected to the board configured as the back-to-back clock master must be designated at the CT-Bus master.
Setting up Channels on the Triton T1/E1 Board
This section discusses setting up T1 CAS, T1 PRI, E1 CAS, or E1 PRI channels on the Triton T1/E1 board.
Configuration window, shown below. The Triton T1/E1 Board can be configured to either CAS or PRI through the configuration options in the window.
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The CH -> Type list on the left side of the window displays the channel types.
Note: In a tie-trunk configuration, set the trunks to “Out of Service” before changing the trunk type from T1 to PRI or vice versa. Otherwise, the system will generate garbage call records to your internal or
external logger service. See “Setting General Trunk Attributes” on page 137 for details.
Figure 59. T1 PRI Protocol Configuration dialog box (top half)
Figure 60. T1 CAS Protocol Configuration dialog box (top half)
Selecting Span Types
•
T1 CAS – Select this option to associate all channels on the span to T1 CAS.
•
Regular ISDN PRI – Select this option to indicate 23B+D ISDN PRI span and to designate the last channel as the D channel.
•
Enable Tie Trunk – Check this box to enable a tie trunk. Tie trunks must terminate to a system also configured as a tie trunk.
Note: This option not available when E1 CAS is selected.
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Figure 61. E1 PRI Protocol Configuration dialog box (top half)
Figure 62. E1 CAS Protocol Configuration dialog box (top half)
Selecting Span Types
•
E1 CAS – Select this option to associate all channels on the span to E1 channel associated signaling.
•
Regular ISDN PRI – Select this option to indicate 30B+D ISDN PRI span and to designate the 16th channel as the D channel.
•
Enable Tie Trunk – Check this box to enable a tie trunk. Tie trunks must terminate to a system also configured as a tie trunk.
Note: This option not available when T1 CAS is selected.
Setting the ISDN PRI Switch Mode
If you select a Span Type of Regular ISDN PRI in the T1 PRI Configuration Window, use the following guidelines to set the ISDN PRI Switch mode.
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Figure 63. T1 PRI Switch Mode
The top four settings are used for a connection to a CO switch:
•
AT&T 4ESS PRI
•
AT&T 5ESS PRI
•
NT DMS-100 PRI
•
NI-2 PRI (default)
The bottom four settings are used for a PRI tie trunk configuration where two MAXCS systems are connected back to back. In such a configuration, one MAXCS system must be configured as Network and the other as
User. For example, set one to NI-2 PRI Network and the other to NI-2 PRI.
•
AT&T 4ESS PRI Network
•
AT&T 5ESS PRI Network
•
NT DMS-100 PRI Network
•
NI-2 PRI Network
If you select a Span Type of Regular ISDN PRI in the E1 PRI Configuration window, use the following guidelines to set the ISDN PRI Switch mode.
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E1 PRI
Figure 64. E1 PRI Switch Mode
The top three settings are used for a connection to a CO switch:
•
Austel TS014 PRI
•
ETSI NET5PRI
•
NT DMS-100 PRI
The bottom three settings are used for a PRI tie trunk configuration where two MAXCS systems are connected back to back. In such a configuration, one MAXCS system must be configured as Network and the other as
User. For example, set one to NT DMS-100 PRI Network and the other to NT DMS-100 PRI.
•
Austel TS014 PRI Network
•
ETSI NET5PRI Network
•
NT DMS-100 PRI Network
Configuring an ISDN Numbering Plan
The ISDN Numbering Plan button in the Protocol Configuration window opens the PRI ISDN Numbering
Plan dialog box. This function allows you to select how the system will identify and code the Called Number for six different types of calls. This coding instructs the CO on how to interpret the number being sent to it.
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Figure 65. PRI ISDN Numbering Plan dialog box
The PRI ISDN Numbering Plan dialog box displays the six classes of numbers (call type) that can be sent to a CO:
•
101CCCC Numbers – CIC (Carrier Identification Codes) dialing.
•
011 International Numbers – For placing calls outside the U.S.
•
1+10-digit Numbers – For local and long distance calls that require dialing 1 before the number.
•
10-digit Numbers – For local and long distance calls that do not require 1 before dialing.
•
7-digit Numbers – For calls placed within the local area that do not require an area code or a 1 prefix.
•
All Other Numbers – For calls that do not fall into any category above, for example, 911, 311.
For each class, select the type of number/numbering plan from the list:
•
Type of Number:
–
Unknown
–
International
–
National
–
Network Specific
–
Subscriber Number
•
Numbering Plan:
–
Unknown
–
ISDN
–
National
–
Private
The setting Unknown is used when the user or network has no knowledge of the numbering plan. In this case, the number digits field is organized according to the network dialing plan.
B Channel Maintenance Message:
This setting controls B channel initialization and maintenance message exchange between MAXCS and the
CO, when the system starts up. Select the maintenance message that will be delivered on the B Channel:
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•
None – No maintenance message sent; puts channel in ready state automatically.
•
Restart – Only sends RESTART message; puts channel in ready state when RESTART ACK
(acknowledgement) response is received from CO.
•
Service – Only sends SERVICE message; puts channel in ready state when SERVICE ACK
(acknowledgement) response is received from CO.
•
Restart and Service – (The default setting) sends both RESTART and SERVICE message; puts channel in ready state when RESTART ACK and SERVICE ACK is received from CO.
Enable PRI Caller Name – Check this to enable PRI caller name
Setting the NSF
The NSF (Network-Specific Facilities) is used with PRI to instruct the CO to route a call to a specific carrier or long distance service. Use the list to identify the type of carrier service you want to use for your ISDN PRI lines.
The choices in the list depend on the specific switch and your long distance service provider. An example of such service includes AT&T Megacom.
Note: If your CO requires specific NSF features to be present in the call setup packet, please contact
AltiGen’s Technical Support department with such information from the CO and they will help you configure it.
Setting a TEI
The TEI (Terminal Endpoint Identifier) defines which terminal device is communicating with the CO switch for a given message. PRI messages involve point-to-point configuration in which each side already knows the source of any message received. ISDN messages involve point to multi-point locations in which the source can only be identified by the TEI.
Select one of the following TEI settings:
•
Default setting – This is the recommended setting.
•
Manually set to – This should always be set to 0. Typically, a zero (0) is used for TEI on a PRI connection.
In some cases where a shared D channel is used, other TEI values might be required to identify which span will be used for a call.
•
Assigned by CO switch – Do not use this setting unless advised by your CO.
Setting PRI Calling Numbers
A PRI Calling Number Setting in the bottom half of the Protocol Configuration dialog box lets you set the numbers you want your Carrier to accept.
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Figure 66. PRI Calling Number Setting
Most PRI trunks allow a MAXCS system to send calling numbers. For example, 10 different extensions in the same PBX system have 10 different DID numbers. With the calling number feature provided by Carriers, the callee will receive a more accurate caller ID.
PRI Calling Number can also be used in a mobile extension or IP hop-off to PRI trunk, so the callee can receive a more accurate caller ID.
When a PRI span is subscribed, a block of DID numbers will be provided by the Carrier. The Carrier should be able to accept Calling Numbers in the DID number block. However, if the numbers are not in the blocks or the digit lengths are mismatched, the Carrier might "reject" the call.
The PRI Calling Number Setting addresses this issue. Choose from three options:
•
Carrier can accept anything as Calling Number (default)
•
Carrier can only accept Calling Number with a minimum of n digits
•
Carrier can accept only assigned numbers as the Calling Number.
If you select the third option, specify “assigned numbers” by clicking the Add button and entering the numbers.
To edit or delete a number you added, select it and click the Edit or Delete button.
If MAXCS detects the Calling Number is not accepted by the Carrier, it will always send the number you enter in the text box at the lower right side of the dialog box as the Calling Number. Enter an appropriate Calling
Number in this box.
Installing a Channel Service Unit (CSU)
This section discusses installing a CSU to the Triton T1 or T1/E1 Board. The channel service unit is a device used to connect a digital trunk line coming in from the phone company to the PBX. A CSU can terminate signals, repeat signals, and respond to loopback commands sent from the central office. A CSU is mandatory for connecting to AltiGen’s T1/E1 board.
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1.
Connect the CSU (Adtran model T1 CSU ACE used as an example) to the T1/PRI or T1/E1 PRI board using an RJ-48C or RJ-48X cable.
2.
Connect the CSU to the network termination box using an RJ-48C or RJ-48X cable.
AltiGen T1 Socket (RJ-48)
Pin 1=Receive Ring (INPUT)
Pin 2=Receive Tip (INPUT)
Pin 4=Transmit Ring (OUTPUT)
Pin 5=Transmit Tip (OUTPUT)
Refer to your CSU manufacturer’s
manual for the proper pinout.
Note: CSUs also are used for line lengths over 75 feet, which helps to resolve attenuation issues.
Troubleshooting T1/E1 Common Symptoms
The most common problems when installing T1 CAS or T1 PRI services:
1.
The service provider misconfigures your T1 CAS/T1 PRI service or terminates your service improperly.
2.
T1 is installed but not turned on because there is no termination device for a period of time.
3.
T1 is turned on but channel is not in service.
E1 Configuration” on page 112.
Configuring Virtual Board SIPSP
A VoIP connection typically consists of two parts.
•
Signal Channel – Responsible for setting up and tearing down a call using protocol. For example, SIP protocol is used in MAXCS to build a signal channel between the server and the IP phone.
•
Media Path – Responsible for encoding, transmitting, and decoding voice for both parties. For example, when an IP phone user makes a call to an outside number, the voice will be encoded at the IP phone, transmitted to the system via the IP network, decoded by the VoIP codec, and passed to a trunk port so that the external party will hear the voice.
The purpose of the virtual boards SIPSP is to build signal channels for different connection types, IP extensions, SIP Tie Trunks, and SIP Trunking from ITSP. Each channel will have its channel ID similar to channels on a Triton extension or trunk board. When an IP phone registers to the system, a channel ID will be assigned to the IP extension. However, these channels are only responsible for processing protocol and call control signals. They require a media path from a VoIP board or from the IP phone to establish a voice steam so that both sides can hear.
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Notes:
•
Make sure you have enough VoIP resource boards.
•
The more signal channels, the more system memory and CPU power required. Proper planning is essential.
•
Changing the number of signal channels requires that you stop and restart the switching and gateway services.
•
SIP Trunking Channel requires a license to activate.
Configuring the SIPSP Board
Double-clicking a SIPSP board in Boards view and then clicking the Board Configuration button opens the
SIP Signaling Channel Configuration panel.
Figure 67. The SIP Signalling Channel Configuration panel
MAXCS is set by default to support 60 SIP extension channels. You can change the number of SIP extension channels and tie-trunk channels. The maximum number possible depends of the system CPU performance, call volume, and usage. If a high performance machine is used as the Softswitch server, the number of channels can be more than 1000. If you change the numbers in this dialog box, you must shut down and restart the switching and gateway services for this change to take effect. When the services restart, the new configuration appears in the Currently Configured Channels fields.
Note: If you change the number of SIP extension or tie trunk channels, you must stop and restart the switching and gateway services.
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Click Advanced Configuration to manage the Trusted SIP Device list.
To move an IP address from one list to the other, select the IP address and click either the right or left arrow button.
Figure 68. SIPSP Advanced Configuration panel
Block Unauthorized SIP Invite Messages
You can block unauthorized SIP invite messages by selecting a checkbox. This setting is disabled by default.
When you enable this setting, the SIP Invite requests are ignored if IP address is not configured in one of the following places:
•
SIP trunk
•
IP dialing Table
•
Trusted SIP Device List
IP extensions for the following devices are added to the Trusted SIP Device list automatically, once they successfully register to the system (unless they are found in the Malicious SIP Device list:
•
AltiGen IP Phones
•
Third-Party SIP Devices
•
IPTalk
To enable protection,
1.
In MAXCS Administrator, open Boards view and double-click SIPSP.
2.
Select Board Configuration > Advanced Configuration.
3.
Check Block Unauthorized SIP Invite and click OK.
Figure 69. Block Unauthorized SIP Invite checkbox
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Fax Routing
To simplify administrative tasks, you can allow voice and fax calls to run on the same SIP trunk channel. This feature applies only to systems using AltiGen SIP trunks. The trunks must be configured to support both voice and fax.
The SIP trunk uses the same SIP server IP address, but different authentication credentials for voice trunk versus a fax trunk.
Figure 70. The Enable Fax Trunk Routing checkbox
Configure Out Call routing just as voice and fax are supported in the same SIP trunk assigned with this SIP
Trunk Profile. Outbound calls made through SIP channels configured for fax channels are for fax only. Hence they should not be assigned trunk access codes or be included in the out call routing for voice calls.
For AltiGen SIP trunks, you must configure one SIP trunk channel to perform SIP registration for GW1 and
GW2 of voice trunk and GW1 and GW2 of the fax trunk individually.
Configuring Virtual Board HMCP
This section is for a gateway Softswitch with an HMCP media server installation only. A single all-in-one system does not require configuration of this board.
Host Media Control Processing (HMCP) is a virtual board that uses an Intel CPU to provide the following functions:
1.
Process VoIP Media Stream
–
Encode, decode, and transcode voice stream
–
Detect and generate tone for IP devices
–
Play music when device is on hold
2.
Play and Record Voice Files
•
Announce system and queue phrases
–
Process auto attendant
–
Process voice mail
–
Call recording for IP extensions
3.
Provide Conferencing Resources
–
Station conference
–
Meet-Me conference
–
Barge-in/silent monitor/coaching
From a deployment point of view, an HMCP media server can be installed in the same Softswitch system sharing the same CPU or can be in a stand-alone server with a dedicated CPU.
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In MAXCS, the HMCP system has been redesigned to perform load balancing on a multi-core system, in a round-robin method. This results in a more even distribution of work across all channels, resulting in better more consistent performance.
Notes
•
Do not install HMCP service in a system with AltiGen's Triton telephony board. It will cause resource conflict.
•
Remove the Triton Resource board and MeetMe conference board from OFFICE systems running as a gateway.
•
An HMCP Media Server license is required to activate an HMCP virtual board.
Figure 71. Example of HMCP licenses in the License Information list
By default the system grants 60 conference members in a maximum of 40 bridges.
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You can change the number to as many as 120 members in a maximum of 40 bridges, and you can activate other HMCP resources, by double-clicking an HMCP board in Boards view and then clicking Board
Configuration.
If you decrease the number of
HMCP resources, the system must be rebooted for the configuration to take effect.
If you increase the number of resources, the system does not have to be rebooted.
Figure 72. The HMCP Board Configuration window
You may change the assigned number by entering a different number (up to the number your system is licensed for and not to exceed the maximum limit for each HMCP board) in the Assigned to this board fields and clicking Apply.
HMCP Resources – Shows the total number licensed (if applicable), total currently assigned, and the number assigned to this HMCP board for the following resource types:
•
Voice Processing Resources (VPR)
•
Video Forwarding Resources (for Polycom VVX phones) – You must assign sufficient video forwarding resources. Be aware that the additional resources you assign may result in a small performance decrease.
Refer to the MaxCS 8.0 Polycom Configuration Guide for details.
•
Station Conference Members
•
MeetMe Conference Members
•
Agent Supervision Bridges
The maximum number of resources that can be supported on an HMCP virtual board is as follows:
•
G.711 VPR — 1,000
•
G.711 / G.722 / G.723 / G.729 VPR
•
Station Conference Members — 120
•
MeetMe Conference Members — 120
•
Agent Supervision Bridges — 20
Notes
•
Codec G.722 is part of a combo codec and is controlled by license.
•
When adding additional combo licenses, the system will also increase the RTP ports it uses and will use these new ports. If these additional ports are not added to the firewall ,calls will not have audio.
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•
1,000 G.711 voice processing resources will be licensed to the system when one HMCP Media Server license is registered.
•
The more VPR assigned, the slower the system will be when it starts up. To calculate the optimized number of VPR you need, use the following formula:
Total G.711 VPR = Total number of extensions X 2
Total G.711/G.722/G.723/729 VPR = Total number of remote IP phone users + Total Tie Trunk Channels that will use compressed codec
•
Adding HMCP licenses or changing assigned numbers does not require restarting the AltiGen switching service.
•
In the event that you need to decrease the assigned numbers of HMCP resources (reassigned to the second HMCP server, for example), the system must be rebooted for the configuration to take effect.
Parameters in IP Header - QoS and TTL assignments.
QoS assignment – IP TOS/DiffServ Byte Value. The default TOS/DiffServ byte hex value "A0"
(10100000) signals the network switch and router that RTP packets are "Critical". To set the value for
Diffserv Code Expedited Forwarding (DSCP EF), you can enter hex value "B8" (10111000).
TTL assignment – For IP paging multicasting only. The purpose of the TTL (Time To Live) is to regulate how many hosts the IP paging packets can pass through. The TTL value is reduced by one on every hop.
You may need to adjust this value if there are remote AltiGen IP phones at different locations that register to MAXCS through WAN and require the IP paging feature. The value will be the number of routers from
MAXCS to remote IP phone plus one.
Media Pass-Through Support for HMCP
One feature in MAXCS for system performance and voice quality is the Media Pass-through feature. This feature applies only to HMCP Softswitch systems; it is enabled by default.
While the Media Pass-through feature is enabled, the HMCP driver doesn’t need to do encoding and decoding on both channels. This approach has the following benefits:
•
Improved system performance, because no MIPS is required for codec processing
•
Improved voice quality, because no distortion is introduced by additional compressed codecs
•
Reduced voice latency, because it eliminates the delay introduced by codec and jitter buffer processing
In addition to benefits to direct calls, pass-through applies during call recording, silent monitoring, and coaching.
G.722 pass-through is enabled by default.
Conditions Not Supported
Media pass-through cannot support all conditions in the HMCP system, even if both connected channels are using same codec. The following MAXCS features do not supported media pass-through:
•
Call playing – The trunk call playing must use encoder and decoder on RTP channel
•
Conference calls – All RTP channels in a conference bridge must be encoded and decoded voice
Assign HMCP Resources to IP Extensions
After you configure the HMCP board, you need to configure extensions to use the HMCP voice processing, conferencing, and recording resources.
In Extension Configuration > General > IP Extension panel, change the Home Media Server ID to the
HMCP Media Server ID if necessary. Please refer to the following scenarios.
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Scenario: Single Standalone HMCP Media Server
For 200 to 1,000 users without an extensive amount of recording resources, fewer than 200 concurrent recording sessions, deploy a stand-alone HMCP Media server as shown below.
The Home Media Server ID should be changed to "01" for all IP extensions, assuming HMCP Media server is using ID 01.
HMCP Codec Preference
MAXCS includes a feature that is designed to help you reduce the CPU consumption that occurs are a result of codec encoding and decoding. It allows you to indicate a codec preference (G.729 or G.711) for calls handled via a SIP trunk. This approach eliminates the steps of encoding and decoding - packets are directly forwarded to the endpoint.
Note: Hardware chassis and hardware gateway configurations do not support the Codec Preference option.
The preference that you set must be supported by the SIP Trunk provider, and must be included in the codec profile list for SIP Trunks.
While this feature is enabled, then the SIP device's codec configuration in Enterprise Manager will be ignored.
Codec preference applies to HMCP only.
All SIP devices must support G.711 uLaw; if there are no common codecs on the device side, and then G.711 uLaw will be used.
This codec preference applies to all SIP Trunk inbound calls and direct extension to SIP trunk outbound calls that are initiated from a phone or dialed via a MAXCS client.
Fax-over IP overrides this setting; it will always use G.711.
Outbound calls initiated by the system (such as calls from the voicemail system or ONA )may not use the preferred codec.
To configure this feature,
1.
Log into MAXCS Administrator with the superpassword. If you do not log in via the superpassword, the feature will not be enabled.
2.
Open the HMCP board configuration panel (double-click HMCP in the Boards window).
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3.
To enable the feature, select the checkbox and specify which codec to use (G.729 or G.711 Mu-Law).
Click Ok.
Figure 73. The Codec Preference setting
Codec Preference - Incoming Calls
When the Codec Preference feature is enabled, the system uses the following logic for handling incoming calls, based upon the content of the first SIP INVITE request.
Preferred Codec G.729
First SIP Invite Request Content Codec Used for the Call
SIP Trunk supports G.729
SIP Trunk does not support G.729
The IP Codec table of Enterprise Manager is ignored. The system uses codec G.729 to negotiate with the endpoint.
No codec is enforced.
Preferred Codec G.711 Mu-Law
First SIP Invite Request Content Codec Used for the Call
SIP Trunk supports G.711 Mu-Law
SIP Trunk does not support G.711 Mu-Law
The IP Codec table of Enterprise Manager is ignored. The system uses codec G.711 Mu-Law to negotiate with the endpoint.
No codec is enforced.
If the target extension or DNIS number is enabled for Fax-Over-IP (FoIP), then no codec is enforced.
Codec Preference - Outgoing Calls, Third-Party IP Phones
When the Codec Preference feature is enabled, the system uses the following logic for handling outgoing SIP trunk calls from third-party IP phones. (Calls are considered SIP trunk calls if the target has a SIP trunk access code or an outcall routing access code prefix.)
Preferred Codec G.729
First SIP Invite Request Content Codec Used for the Call
Supports G.729
Does not support G.729
The IP Codec table of Enterprise Manager is ignored. The system uses codec G.729 to negotiate with the endpoint.
No codec is enforced.
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Preferred Codec G.711 Mu-Law
First SIP Invite Request Content Codec Used for the Call
Supports G.711
Does not support G.711
The IP Codec table of Enterprise Manager is ignored. The system uses codec G.711 to negotiate with the endpoint.
No codec is enforced.
If the source extension is enabled for Fax-Over-IP (FoIP), then no codec is enforced.
Codec Preference - Outgoing Calls, IP Phones
Because IP Phone SIP call requests are always from MAXCS, the behavior is different from calls from thirdparty phone extensions.
When a user makes an outgoing call from an IP phone, the system follows the IP Codec table in Enterprise
Manager. All IP phones support G.729 and G.711 Mu-Law.
When the Codec Preference feature is enabled, the system uses the following logic for handling outgoing SIP trunk calls.
G.729
Preferred Codec
G.711 Mu-Law
Codec Used for the Call
The system modifies the IP phone's codec in the RE-INVTE SDP body with
G.729 codec. After the modification, MAXCS must pick G.729 and G.711 codecs as the preferred codecs to negotiate with the endpoint.
The system modifies the IP phone's codec in the RE-INVTE SDP body with
G.711 codec. After the modification, MAXCS must pick G.711 and G.729 codecs as the preferred codecs to negotiate with the endpoint.
If the source extension is enabled for Fax-Over-IP (FoIP), then no codec is enforced.
Configuring the MAX1000/2000 Board
The MAX1000/2000 Server is a telecom appliance that consists of an embedded DSP board and two access board slots. MAXCS treats the entire MAX system as one board with two access board options. The Boards window displays the name of the MAX board, followed by [xxyy(-T1),xxyy]:
Figure 74. Boards View showing MAX board
xx refers to the number of analog trunks, and yy
refers to the number of analog extensions. If an access board has a T1/E1 port,
-T1
is added to the end.
In the Boards window, double-click the MAX 1000/2000 board to open the main Board Configuration window.
The Channel Group Info panel shows the channel groups (groups of channels that belong to the same type).
For example, if one 4x4xT1 access board and one 4x8 access board are installed in the MAX 1000/2000 main board, there will be three channel groups for the 4x4xT1 card, and two channel groups for the 4x8. When one of the channel groups is selected, the Channel Mapping List reflects the selection.
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•
Double-clicking a T1/E1 channel group opens the channel group configuration dialog box. For information
E1 channel groups only.
•
In the channel group configuration dialog box, click the Protocol button to open the Protocol
Figure 75. Board Configuration window
Double-clicking a channel in the Channel Mapping List opens the appropriate configuration dialog box for that channel.
•
For details on configuring the T1/E1 trunk, see “Triton T1/E1 Trunk Properties” on page 152.
•
•
the Board Configuration button opens a dialog box.
Figure 76. MAX 1000/2000 Board Configuration window
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This dialog box displays the board serial number, top access card serial number, bottom access card serial number, DSP clock, board ID, physical ID, and logical ID. You can choose to configure the board as either T1 or E1, then click OK. Additional steps are needed to further configure the CAS or PRI protocol in the Protocol
Configuration window, shown in (see Figure 59) and (see Figure 60).
Note: Hardware chassis such as the Max1000 do not support the G.722 codec, the media pass-though feature, or the T.38 feature.
Configuring the Virtual MobileExtSP Board
A simulated physical board – MobileExtSP board – is created in the Softswitch server when you install the
MAXCS system. This single MobileExtSP board handles all mobile extensions.
Configuring the virtual MobileExtSP board is discussed on page 230 in the chapter “Mobile Extension
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H A P T E R
11
Trunk Configuration
Trunk attributes and parameters are set using the Trunk Configuration window. The attributes and options available depend on the type of board and trunk. This chapter discusses general configuration options applicable to all trunks, followed by specific configuration options for the following trunk types:
•
•
•
•
configurable on tabs in the Trunk Configuration window.
Trunks Out of Service
If none of the trunks are available when an outside call is placed, the caller will hear the system prompt: “All outside lines are busy. Please try again later.”
Channel Identification
Channel Physical Location. The Channel Information panel shows logical board ID, board name, channel group type, and channel ID:
Figure 77. Channel Information box
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Opening the Trunk Configuration Window
To open the general Trunk Configuration window, do one of the following:
•
Click the Trunk Configuration button in the toolbar.
•
Select PBX > Trunk Configuration.
•
Double-click a trunk in the Trunk View window.
Selecting Channel
Properties from the right-click menu in
Trunk View bypasses the general Trunk
Configuration window to open a trunk properties window specific to the selected trunk.
Figure 78. Trunk View window
The Trunk Configuration window opens.
Figure 79. Trunk Configuration, General tab
Selecting Trunks to Set Attributes
The title bar of the Trunk Configuration window displays the card and the channel of the selected trunk.
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Configuring One or Multiple Trunks
The list on the left shows all the configured trunks. The Location format is the same as in the Trunk View window, that is, Logical Board ID : Channel Number. The logical board ID is assigned by the system. This ID may change when a telephony board is added into or removed from the system.
When you select a trunk in this list, the options and parameters for the trunk appear in the settings in the right side of the window.
Configuring One or Multiple Trunks
To customize trunk characteristics, you work on one trunk at a time. To apply the same configuration to multiple trunks, use the Apply To button. This opens a list of all trunks, with all of the trunks selected by default. Select the trunks you want to apply changes to, then click OK. (Use Ctrl+click and Shift+click to select several trunks.) This applies changes to multiple trunks for only the attribute or option that you changed.
Figure 80. The Trunk Selection window
Setting General Trunk Attributes
Select a channel to view its current attributes. You can then set or change the following attributes. If an option is grayed out, it is not available for that type of trunk:
•
Access Code – Assign a trunk access code to the selected trunk. If you need to use a trunk access code
System Number Plan” on page 52).
Note: There are two types of access code: Trunk Access Code (TAC) and Route Access Code
(RAC). TAC is a quick and easy way to select which trunk(s) you would like to dial out from, especially when you want to reserve trunks for a special dialing purpose. For example, you can set up TAC "7" and assign that to trunk(s). These trunks will be reserved exclusively for users who know the TAC "7".
Although TAC is easy to use, it does have limitations especially when you are located in an area with a complicated dialing pattern or you need to set up VoIP hop-off dialing.
RAC uses the Out Call Routing table, which has the flexibility to group trunks into a route, assign routes to a specific dialing pattern, and add/delete digits from the dialing pattern. It can solve most of the complicated dialing problems. If your system is using RAC, you can set this
TAC field to "None".
•
Area Code – The local area code for each trunk. Enter a three-digit area code. If left blank, the trunk assumes the home area code defined in the General tab of the System Configuration window. This
configuration is for each trunk in the system and will negatively affect features such as Zoomerang if the area code is not configured prope
rly.
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•
Direction – The trunk direction can be Outgoing only, Incoming only, Both Outgoing and Incoming,
Paging, or E911. The Both option is the system default.
Note: If a trunk is in the hunt group of your company main number and you configure this trunk as an "Outgoing" trunk, the incoming call will be rejected by the system. To avoid this mistake, make sure you check with your carrier to verify the hunting number before you configure a trunk to Outgoing.
Paging – This configuration is for an overhead paging device and requires a Loop Start trunk port. The paging equipment will provide loop current to the trunk port.
When this option is selected, you can assign an ID in the list. The range of paging IDs are from 00 to
99, which allows MAXCS to be connected to up to 100 paging systems through trunks for multi-zone paging applications.
To activate a trunk paging port, dial #45 and the ID number. For example, a user dials #4508 to connect to a paging system through the trunk with paging ID of 08.
Audio Out port on the telephony board and uses #44 to activate.
The E911 option is exclusively for an analog Centralized Automatic Message Accounting (CAMA) trunk connecting to a Triton analog trunk board. CAMA trunk is a special type of trunk from your carrier for
E911 service. When an analog trunk port is assigned as an E911 CAMA trunk, the system will send the station identification number, defined in the extension configuration E911 CID field, to the PSAP via multi-frequency signaling. The E911 CID is needed to:
•
Allow PSAP to identify the caller's information and exact location by matching the Automatic Location
Identifier database in PSAP.
•
Have the callback number in case the call is disconnected.
Note: Do not select the E911 option for a T1-CAS or PRI trunk. T1-CAS cannot transmit the
ID. PRI trunk will transmit calling party's ID automatically. When the E911 option is checked, this trunk will no longer receive inbound calls, and only 911 calls will go out through this trunk. Each state may have different E911 regulations and requirements.
Please check with the local authority to understand what is required by law.
•
Phone Number – If this trunk is an analog or T1-CAS trunk, this field is used for labeling purposes only.
Enter the number without area code in this field. If this trunk is a PRI trunk, the system will output this number to the carrier as the calling party CallerID.
PRI trunk transmitting caller ID rules:
1.
If extension has Transmitted CID configured, this number will be transmitted first. If not configured, go to next.
2.
If extension has DID Number configured, the 10-digit DID number will be transmitted. If not configured, go to next.
3.
If PRI trunk channel has area code and caller ID configured, this number will be transmitted.
If not configured, go to next.
4.
PRI will transmit the system home area code and main number defined in System
Configuration, General tab.
•
Description – Descriptive information such as the company name for the assigned Phone Number, or appropriate agency if this trunk provides 911 access.
•
Trunk Dialing Scheme – Overlap or En-bloc dialing.
•
Overlap – Transmitting dialed DTMF digits to the CO without buffering digits in the system first. Use
Overlap dialing for analog and T1-CAS trunks for best results. Calls will be completed faster.
•
En-bloc – The system will buffer all dialed digits and send it to the CO at once. Typically is used in
ISDN-PRI trunk and SIP trunk.
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Note: For IP tie trunks, use the IP Dialing Table in Enterprise Manager to set the dialing scheme (Enterprise Manager is available by selecting VoIP > Enterprise Network
Management, or from the Windows Start menu).
•
Trunk Call Predial String – To have the system automatically insert the configured digits whenever the selected trunk is used for outgoing calls. This feature is used to prevent having to dial “9” twice for trunk access when the system is used behind another PBX system or this trunk is a Centrex line, which requires dialing “9” to make a call. If you select this option, type the predial digit(s) into the text box.
•
Enable Centrex Transfer – When checked, the system is able to transfer an incoming call to another outside number through the same trunk and release the incoming trunk. Before you configure this option for the trunk, please make sure your trunk is a Centrex line or supports the Release Line Transfer (RLT) feature. Depending on the type of trunk, your configuration may be different:
•
If this is an analog Centrex line, you only need to check the Enable Centrex Transfer check box. A
FLASH signal will be transmitted to the CO if the incoming trunk call needs to be transferred to an outside number.
•
If this is a T1-CAS trunk, you may need to add “transfer predial string.” From the CO point of view, it is their feature code to initiate RLT. Please check with your carrier to get the specification.
•
If this is a PRI trunk, you need to ask your carrier if they support RLT through DTMF. Some carriers accept *8 to signal RLT. AltiGen PRI trunks currently do not support 2-B channel transfer feature.
How to signal MAXCS that it is a Centrex transfer:
•
If a call is connected to an extension, the extension user needs to dial
FLASH * plus trunk access code and the outside number.
•
If a virtual extension forwarding or speed dialing number is configured to an outside number and the extension user transfers a call to the virtual extension or speed dialing number, the system will add the
Centrex FLASH automatically. You don’t need to add the “*” in the forwarding or speed dialing digit stream.
•
Attribute – In Service makes the trunk available for use. Out of Service prevents the trunk from being used (for example, while performing maintenance).
•
Enable Tie Trunk – This configuration field is meaningful only if you use T1 or PRI to connect two MAXCS systems back-to-back. Do not check this box if you connect a MAXCS to a third-party PBX via T1 or PRI trunk.
When this configuration is checked, the system software will interpret the incoming [ANI] [DNIS] digit sequence as [Caller’s Extension Number] and [Target Extension Number]. An incoming tie trunk call will be routed to the target extension and all the Incall Routing rules will be bypassed. If you do not check this box for system-to-system tie trunk, the system will check the Ext. DID/DNIS Routing/Caller ID Routing table first. If there is no match, the trunk incall routing rule will apply.
Note: The Enable Tie Trunk field under Board Configuration > Protocol needs to be enabled for T1/PRI tie trunks as well. It will tell the system to transmit [Caller’s Extension
Number] and [Target’s Extension Number] as [ANI] [DNIS] to the other system. In case this is a T1-CAS, which typically cannot transmit any data to the CO, the system will use
DTMF as a way to transmit [Caller’s Extension Number] and [Target’s Extension
Number] to the other side of the tie trunk. Because the format is AltiGen proprietary, you may have a problem if you enable this configuration when connecting to a non-AltiGen
PBX.
•
Holiday Profile – A holiday profile can be assigned to a trunk. The list selection is based on settings
configured in the Holiday tab of System Configuration (see “Routing Calls on Holidays” on page 58).
•
Business Hour Profile – A business hour profile can be assigned to a trunk. The list selection is based on settings configured in the Business Hours tab of System Configuration.
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•
Recording Option – Recording for incoming and outgoing calls is supported for Triton Analog, T1/E1, and
IP trunks; use the list to select Disable or Enable. If you select Enable, choose the license you want to assign (Concurrent Session or Dedicated Seat), and make sure that in System > Recording
Configuration one of the trunk-based recording options is selected.
Note: When you use trunk-based recording, inbound or outbound calls are recorded as long as the trunk is in use. For example, an inbound call that is answered by an AA, routed to an operator, and transferred to an extension will begin recording when the AA answers the call and end recording when the trunk is released.
With extension recording, recording starts only when the extension user answers the call.
•
Trunk Properties – Opens a dialog box that allows you to configure low-level, hardware-specific properties for each trunk. The options vary depending on the type of board and trunk; this is discussed in subsequent sections.
SIP Tie Trunk Properties
To open a configuration dialog box for a SIP tie-trunk channel, do one of the following:
•
If you’re in the Trunk Configuration window, select a Triton VoIP channel from the trunk channels list, then click the Trunk Properties button, or just double-click the channel in the list.
•
If you’re in the Trunk View window, right-click the channel and select Channel Properties.
Figure 81. Configuration dialog box for a Triton VoIP channel
See “Configuring the SIPSP Board” on page 124 for board configuration information.
Note: This is signal only trunks. Make sure you have enough IP resource boards to cover your needs.
SIP Trunk Properties
Traditionally, telecom trunks are from your local carrier’s PSTN switch and the dial tone is provided via either analog trunks or T1/PRI digital trunks. A new type of service called “IP Dial Tone,” which allows you to dial a long distance call at a lower rate, is available. IP Dial Tone is delivered through your IP data network, and the service provider can be anywhere in the world, as long as the VoIP data packets can be routed properly.
If you have SIP-based IP dial tone service from an Internet Telephony Service Provider (ITSP), you need to configure SIP trunk channels to connect to the service. Before you start, note the following:
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•
An AltiGen SIP Trunking channel is licensed. You need to buy and register a license to be able to configure this option. Note that AltiGen SIP Trunks do not require a SIP Trunk license.
•
Beginning in release 8.0, the number of SIP trunk channels allocated is now based on the configuration.
In earlier releases, this was based upon the number of SIP trunk licenses.
•
AltiGen does not guarantee the voice quality of the SIP dial tone coming from your service provider. You need to work with your data service and SIP trunking service provider to make sure adequate QoS is provisioned for your WAN service.
•
AltiGen does not guarantee SIP trunk implementation will work with all SIP dial tone service providers.
AltiGen dealers are notified of AltiGen-tested and certified SIP-Trunk service providers. Configuration guidelines for each AltiGen-certified SIP-Trunk service provider can be found in the AltiGen authorized
Partner Knowledge Base, available from the AltiGen Partner Web Site. SIP dial tone service providers need to support the following:
•
G.711, G.723.1, G.729 codec
•
RFC 2833 for DTMF tone delivery
•
SIP MD5 authentication with SIP registration
•
If MAXCS is behind NAT, verify that your SIP SP can support this configuration.
When subscribing to a SIP dial tone service, typically your service provider will provide you with the information
to each SIP trunk channel configuration individually.
Note: This is signal only trunks. Make sure you have enough IP resource boards to cover your needs.
Important:
You must add the SIP Trunk service provider’s IP address to the IP Device Range in Enterprise
Manager and select the proper codec profile for this service. See “Assigning Codec Profiles to
IP Addresses” on page 333. Failure to do this step may cause no voice path, even if the SIP
Trunk channel shows the call is connected.
SIP Groups
In previous releases of MAXCS, SIP registration was configured by individual channels. This configuration approach had several disadvantages. For example, configuring multiple channels with the same authentication credentials was inefficient. Setting multiple channels to register to the same server and having to configure only one of them with a registration period required configuring that channel twice. And for some scenarios, such as configuring an AltiGen SIP Trunk fax server, one trunk had to be reserved for fax even if faxes were seldom sent or received.
To streamline SIP registration, AltiGen has implemented SIP Group configuration. The following three panels in the earlier releases have been consolidated:
•
The SIP Trunk Configuration dialog box
•
The Trunk Group Configuration dialog box
•
The SIP Trunk Profile dialog box
The Trunk Configuration dialog box has been updated to show the SIP Trunk group, in the Type column. There are various windows within MAXCS Administrator that will now show the SIP Group for a SIP Server, including
•
Trunk View
•
Trunk Configuration
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Configuring SIP Groups
The SIP Signaling Channel Configuration dialog box is the main starting point for all SIP group and channel configuration. (Double-click a SIPSP board in Boards view and then click Board Configuration to reach this panel.)
Figure 82. SIP Trunk Configuration dialog box and Edit box
Two of the buttons in the earlier releases have been replaced with two new buttons:
•
SIP Group Configuration – This is where you now configure SIP Groups, adding SIP servers and configuring various options for each server.
•
Channel Assignment – This is where you now enable and disable channels, and move channels from one SIP group to another.
In the SIP Signaling Channel Configuration panel, you can do the following:
•
Create new SIP groups
•
Add servers to SIP groups (and remove servers from SIP groups)
•
Change the relative priority of servers in a SIP group
•
Configure various registration and SIP Options for a server
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SIP Group Configuration
To create a new SIP Group,
1.
Double-click a SIPSP board in Boards view and then click Board Configuration > SIP Group
Configuration.
Figure 83. The SIP Group Configuration window
2.
Below the Groups list, click Add.
Figure 84. SIP Group parameters
3.
Enter a name for the group and indicate whether this is an AltiGen SIP Trunk.
Note: Release 8.0 includes license checking for AltiGen SIP trunks. The SIP Server address and registration UID will be validated. Be aware that trunks will show as not registered if you start MAXCS Administrator before MAXCS is fully up, because license information will not be ready until then. However, after
30 to 90 seconds, the trunk should be show are ready.
4.
(Optional) If this group will handle fax calls, select the option Enable fax trunk routing and enter the appropriate user name and password. This feature is supported on AltiGen SIP trunks only.
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If the extension is a fax extension and Enable fax Trunk routing is checked, that means the SIPSP should use the Fax Username and Fax password of the SIP Trunk instead of regular username and password of SIP Trunk to negotiate with the SIP Trunk.
If the extension is a fax extension but Enable fax trunk routing is unchecked, that means the SIPSP should use the regular username and password of the SIP Trunk to negotiate with the SIP Trunk side.
SIP Servers
You can add up to four servers to each SIP group.
Adding SIP Servers to a SIP Group
You can add up to four servers to each SIP group. To add a SIP server to a SIP group,
1.
Double-click a SIPSP board in Boards view and then click Board Configuration > SIP Group
Configuration.
2.
Select the appropriate SIP group in the Groups list.
3.
Below the SIP Servers list, click Add.
Figure 85. Add a SIP Server to a group
4.
Enter the domain name.
5.
(Optional) If you want to copy the settings from another server, choose the group and the server from those two pulldown menus.
Note: In the SIP Group Configuration panel, click Refresh when you want to update the view
Removing Servers from a SIP Group
To remove a server from a SIP group,
1.
Double-click an SIPSP board in Boards view and then click Board Configuration.
2.
Select the appropriate SIP group in the Groups list.
3.
Select the server that you want to remove from the SIP group.
4.
Click Del (Delete).
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Changing the Order of Servers in a SIP Group
The order of the servers in the Servers list determines how to severs are accessed. The first server is always used; when that server is not available, the second server in the list is use, and so on.
To change the order of SIP servers within a SIP Group,
1.
Double-click a SIPSP board in Boards view and then click Board Configuration.
2.
Select the appropriate SIP group in the Groups list.
3.
In the Servers list, select the server that you want to move.
4.
Click Up or Down as appropriate.
SIP Server Registration Parameters
The Register tab contains all registration-related settings for a SIP server. All of the settings on the SIP Trunk
Configuration
dialog box from earlier releases of MAXCS are found on this tab.
Figure 86. The SIP Server Register tab
The lone exception is the earlier Enable Channel option, which in Release 8.0 is found in the Channel
Assignment
panel.
SIP Register Parameter
Domain
SIP Server IP Address
User Name
Password
Register Period
SIP Source Port (Non-TLS)
SIP Destination Port
Description
The Domain Name of the SIP Trunk service provider, if required. If there is no domain, you may make the domain the same as the SIP
Server IP address.
The SIP Trunk service provider’s server IP address.
This is assigned by the SIP Trunk service provider.
This is assigned by the SIP Trunk service provider.
How frequently the AltiGen system needs to send SIP registration packets to the service provider. This can detect if the service provider is up or not.
Some service providers do not accept SIP Register messages. In these cases, you can disable sending SIP Register messages from MAXCS by setting the SIP Register Period to 0.
For SIP UDP, select the source port from 5060 or 10060.
For TCP or TLS, you cannot change ports. Using a port other than 5060 will prevent SIP-ALG firewall/router from changing the SIP packets.
A SIP Trunk can have different source port and destination port.
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Note: SIP Server registration status depends upon both the registration results and the SIP OPTIONS results. If either of those two processes fails, the server's status will be set to DOWN. SIP Groups and member channels share the same registration status. As long as at least one SIP server is UP, all of the enabled member channels will show as status IDLE.
SIP Server General Parameters
The Settings tab contains the parameters from the SIP Trunk Profile tab in earlier releases of MAXCS.
Figure 87. The SIP Server Settings tab
SIP Server General
Parameter
IP Address
SIP Protocol Field
Custom P-Asserted-ID
Description
Not Sent (default) – Do not send transmitted caller ID
FROM Header – Send the caller ID using the SIP FROM header
P-Preferred Identity – Send the caller ID using the SIP
P-Preferred Identity header
P-Asserted Identity – Send the caller ID using the SIP
P-Asserted Identity header
This field is designed for support of Caller ID on Verizon SIP Trunks.
Unless you are instructed by your service provider to set these values, you should only use these fields when you connect to Verizon SIP
Trunks.
This field allows you to insert a header into the SIP packet. When this field is not empty, the specified header will be included in the SIP packet.
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Carrier can accept any number
Carrier can only accept Calling
Number with minimum x digits
Carrier can only accept assigned numbers as Calling
Number
Parameter
Custom Diversion
Send Caller Name
Enable Standard Record-Route
Header
Enable SIP REFER
Enable Centrex Transfer
Incoming DID Number Field
Description
This field is designed for support of Caller ID on Verizon SIP Trunks.
Unless you are instructed by your service provider to set these values, you should only use these fields when you connect to Verizon SIP
Trunks.
When this field is not empty, its content will be included in the SIP packet.
If the field does not contain a semicolon, a suffix will be attached, as shown in red in the following line:
Diversion: Custom diversion text; reason=unknown;privacy="off"
If there is a semicolon in the text, the suffix will not be attached, as depicted in the following line:
Diversion: Custom diversion text
This is the default.
Enter the number of digits, then enter a calling number in the field below the table in case the carrier cannot accept configured numbers.
If you select the this option, specify “assigned numbers” by clicking the
Add button and entering the numbers. To edit or delete a number you added, select it and click the Edit or Del button. Enter a calling number in the field below the table in case the carrier cannot accept configured numbers.
Check to also send the caller name to callees.
Check this box if the SIP service provider uses SIP Record-Route and the SIP trunk cannot make or receive calls. If it already works, DO NOT
CHECK or UNCHECK this box. [Service provider Bandwidth.com with
Edgewater Route require this checked]
This option, in conjunction with the Enable Centrex Transfer option, instructs the SIP Trunk provider to release both the inbound and the outbound legs of a transferred call, once the transfer has been completed.
The extension-specific setting for the Release SIP Tie-Link Trunk
feature is discussed in the section Setting Personal Information.
This option is provided to support various gateway devices. For instructions on configuration AudioCodes gateways and other-third party devices, refer to the configuration guides stored in the AltiGen
Knowledgebase.
This option, in conjunction with the Enable SIP Refer option, instructs the SIP Trunk provider to release both the inbound and the outbound legs of a transferred call, once the transfer has been completed.
The extension-specific setting for the Release SIP Tie-Link Trunk
feature is discussed in the section Setting Personal Information.
This option is provided to support various gateway devices. For instructions on configuration AudioCodes gateways and other-third party devices, refer to the configuration guides stored in the AltiGen
Knowledgebase.
When a call comes in, the SIP trunk uses To Header or Request URI as the DID/DNIS number
SIP Server SIP Options Parameters
The SIP OPTIONS tab includes the SIP OPTIONS parameters from the previous releases of MAXCS. The SIP
Server Name parameter is essentially the new Domain field.
You can configure trunk groups so that SIP devices can exchange their status. By sending a “keepalive” message and checking for a valid response, SIP devices will know whether remote peers are ready to receive a new request.
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Once the MAXCS server has been set up, SIP trunks will try to register to the SIP servers first. If the registration is successful and if the SIP Options feature is enabled, the system will send the “keepalive” message to SIP providers according the interval setting. Only one “keepalive” message will be sent for each group of trunks.
Figure 88. The SIP OPTIONS tab
SIP Server registration status depends upon both the registration results and the SIP OPTIONS results. If either of those two processes fails, the server’s status will be set to DOWN. SIP Groups and member channels share the same registration status. As long as at least one SIP server is UP, all of the enabled member channels will show as status IDLE. You can see the status of severs on the SIP Group Configuration panel.
SIP Options can be enabled or disabled for each SIP Server. By default, the SIP Options parameter is disabled.
SIP OPTIONS Parameter
SIP Server IP Address
Enable SIP OPTIONS
SIP OPTIONS Interval
Number of Retries
Retry Interval
Description
Enable or disable this feature for the selected SIP Server.
How often, in seconds, the server sends a “keepalive” message to this enabled SIP trunk group. The default interval is 30 seconds.
If MAXCS receives no 200 (OK) response, the number of times a
“keepalive’ message should be sent. After these retries, if there still has been no valid response, then the system marks all SIP trunks in the group as Not Ready. The default number of attempts is 5.
While a SIP trunk group is in a Retry state and is not receiving a valid response, how often MAXCS should send another “keepalive” message to the SIP server. The default interval is 2 seconds.
If the system does not receive a SIP 200 (OK) message after the set number of retries, it then sets all SIP trunks in that group to Not Ready.
Channels
To view a list of channels and determine the SIP Group to which each channel belongs,
1.
Double-click a SIPSP board in Boards view and then click Board Configuration.
2.
Click Channel Assignment. The panel shows a list of the channels, indicating their status and SIP Group.
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Figure 89. The Channel Assignment panel
A checkbox indicates whether a channel is enabled.
To disable or enable a channel, check or clear its checkbox. To enable or disable all channels, check or clear the checkbox in the column heading at the top of the list.
If you discover that you need to add more channels, close this panel, click Board Configuration, and set an
appropriate value for the Change Number of SIP Trunk Channels to… option (see Figure 82, “SIP Trunk
Configuration dialog box and Edit box"). Then restart the system.
Assign Channels to a SIP Group
To assign a channel to a SIP Group, or reassign it to a different SIP Group,
1.
Double-click a SIPSP board in Boards view and then click Board Configuration.
2.
Click Channel Assignment.
3.
Select one or more channels in the list. Use Ctrl-Click to select multiple channels.
4.
Click Assign Group.
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Figure 90. Assign channels to a SIP Group
5.
Choose a SIP Group from the list and click OK.
Considerations
•
A channel is not enabled if it is not assigned to any SIP group.
•
If you assign a disabled channel to a SIP group, doing so does not automatically enable it - you must manually enable the channel (see the previous section).
•
In order to assign channels to a SIP group, the SIP group must have at least one SIP Server assigned to it.
•
If you remove all SIP Servers from a SIP Group, all channels previously assigned to that SIP group automatically will show as 'Not Ready.'
SIP Trunk TLS Support
MAXCS supports TLS/SRTP on SIP Trunks.
MAXCS SIP trunks will not verify a TLS far-end certificate. A self-signed certificate public key is provided under http://maxcs_ip_addr/altigen.crt, if the SIP trunk provider needs to verify the certificate. TLS/SRTP secures the
SIP signal and voice between MAXCS and SIP trunk service provider.
The following devices are supported for TLS:
•
AudioCodes MP-11x
•
AudioCodes Mediant 1000-B PRI / T1
•
AltiGen SIP Trunks
When configuring SIP Trunks, you no longer see the option Automatic NAT Traversal as you did in previous releases. In MAXCS 8.0, SIP Trunks no longer have this parameter. The pre-8.0 option SIP Source Port has been changed to SIP Source Port (Non-TLS).
When configuring SIPTrunk.com SIP Trunks for TLS, the SIP Destination Point field should be set to port 5067.
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Figure 91. The SIP Destination Port field set to 5067 for AltiGen SIP Trunks
Note: Effective in the 8.0 Release, Enterprise Manager has a new option, TLS/SRTP codec setting.
Following is an example of a generic setup for SIP TLS; configuration may differ from one SIP trunk to another.
Refer to the configuration guide for your SIP Trunk for specific details; these guides can be found in the AltiGen
Knowledgebase.
1.
Make sure your SIP gateway or SIP trunk has TLS/SRTP enabled.
2.
In MaxAdministrator, open Trunk view. Double-click a trunk and click Trunk Properties > SIP Group
Configuration.
3.
Add a group, for example, SIPTLSGrp, in the Groups panel.
Click here to add a new SIP
Group for TLS.
Then click here to add a server to that new SIP group.
Figure 92. Add a new SIP Group for TLS; add a server to that new group
4.
Select the new group in the Groups panel. Below the SIP Servers list, click Add and add a server.
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5.
Select the new server in the SIP Servers list and enter the following parameters on the Register tab:
•
Fill in the SIP trunk server IP address, username, and password for the SIP trunk.
•
If the trunk needs to register to the MAXCS server, enter the register period. Otherwise, enter "0".
•
For the SIP Destination Port, use port 5067 for AltiGen SIP trunks. SIP Source Port is not used here. The
SIP source port will always be 5061 if TLS/SRTP is used.
6.
Save the changes.
Note: SIP OPTION over TLS is not supported.
Triton T1/E1 Trunk Properties
To open a configuration dialog box for a Triton T1/E1 channel, do one of the following:
•
If you’re in the Trunk Configuration window, select a Triton T1/E1 channel from the trunk channels list, then click the Trunk Properties button, or just double-click the channel in the list.
•
If you’re in the Trunk View window, right-click the channel and select Channel Properties.
Figure 93. Triton T1 Configuration dialog box
Following are the parameters for the Triton T1 Configuration dialog box.
Parameter Description
T1 robbed-bit signaling
Protocol
You can set Protocol to one of the following:
•
E&M Wink Start (default)
•
E&M Immediate Start
•
Ground Start
•
Loop Start
For signaling from one board to another, only E&M Wink Start is supported. Loop
Start, Ground Start, and E&M Immediate Start protocols cannot be used for interfacing between two boards.
Dialing Delay
Specifies the delay, in milliseconds, after trunk seizure and before digit dialing.
This configuration will slow down the system transmitting digits to the CO by a defined delay to avoid missing digits. Do not change this value unless advised.
Caller ID and DID Collection
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Or
Parameter
Apply to
Description
You can select the maximum time-out delays, in seconds, and the appropriate sequence of symbols to be collected for Caller ID and DID.
Max. seconds before the first digit
Maximum wait time before time-out for the system to identify this digit after either the first ring in ground start or loop start or the wink in wink start. The range is from
1-6 seconds, or None, with a default value of 3 seconds. Do not change this value unless advised.
None means no Caller ID or DID information will be collected. All other options will be grayed out. Use this option to disable Caller ID and DID collection.
Max. seconds between digits
Incoming sequence
Maximum wait time before time-out between two digits. Default value is None. Do
not
change this value unless advised.
Selecting None means the system will only wait for the sequence of digits that are collected within the length of time specified in the Max. seconds before the first digit field.
Select up to five incoming symbols to collect from the Caller ID or DID digits:
•
None
•
#
•
*
•
# or *
•
Caller ID
•
DID/DNIS
Selecting None in any field of the sequence will terminate the sequence and automatically disable subsequent entries in the sequence.
The default sequence is:
“# or *” (and then) “Caller ID” (and then) “# or *” (and then) “DID/DNIS”
Sets up an additional, alternative sequence. You can select another set of up to five incoming symbols to collect.
Not checking any box is equivalent to checking None in the first field.
The default sequence is:
“DID/DNIS”
If appropriate, you can use this button, as described in “Configuring One or
Multiple Trunks” on page 137, to apply the Caller ID Collection to multiple T1
trunks.
Note: In order for back-to-back T1 and tie trunk T1 configurations to perform properly, it is recommended that you use the system’s default incoming call sequences:
Figure 94. Incoming call sequence parameters
Caller ID and DID Incoming Sequence Example
The following is an example of a Caller ID and DID/DNIS incoming sequence window.
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Figure 95. Sample Incoming Sequence window
When a call comes in, the system tries to match the incoming sequence to either the first or second Incoming
Sequence Digit String sequence. If no match is found, no Caller ID or DID digits will be collected.
•
The system waits 3 seconds for the first digit to arrive. If the symbol is a # , it continues with the first sequence. Otherwise, it looks for a match to the first (and only) symbol in the second sequence, the
DID/
DNIS number.
•
For the example, let’s say the system receives the
#
. It then waits 1 second between each digit for the next digit until all digits are received. The * symbol is a delimiter between Caller ID and DID digits.
In this example, the MAXCS ACC/ACM system is expecting either the sequence
#CID*DID
or only
DID
digits for incoming calls. If no match is found for either sequence, no Caller ID or DID digits are collected.
Triton Analog Trunk GS/LS Properties
To open a configuration dialog box for a Triton Analog Trunk GS/LS channel, do one of the following:
•
If you’re in the Trunk Configuration window, select a Triton Analog Trunk GS/LS channel from the trunk channels list, then click the Trunk Properties button, or just double-click the channel in the list.
•
If you’re in the Trunk View window, right-click the channel and select Channel Properties.
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Figure 96. Triton Analog Trunk GS/LS Properties window
Note that you can use Apply to in this dialog box to apply changes to other trunks of the same type.
Triton Parameter
Interface Type
Incoming Ring
Impedance
Match Impedance button
Description
Select the type of trunk that will interface with this trunk channel:
•
Loop Start Trunk
•
Ground Start Trunk
Single – Default setting for North America
Double – For countries using Ring-Ring-Silent type of ring pattern
The resistance of electrical current to alternating current, measured in
Ohms. Impedance occurs when power or signal is transferred from one circuit to another. When a trunk interface impedance is greatly mismatched with the CO analog line, it may result in static noise and echo heard by IP phone users. The system automatically selects the impedance profile that best matches the Triton trunk interface with the
CO. In the rare case where you are not getting the best match, you can disable this feature by checking Disable Impedance Match During
System Startup, and you can set the Impedance manually.
Changes the Impedance setting to the best match for the selected trunk channel, and then measures noise and returned echo with this impedance setting. Results are displayed in the Diagnosis section of the dialog box. The system automatically runs a matching test upon system startup, unless you disable the feature. If later you connect a new analog line to an empty port or replace an existing line, you need to click this button to best match the impedance.
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Triton Parameter
Match Result button
Disable Impedance Match
During System Startup
Caller ID Receiving
Centrex Flash Duration (ms)
Out of Service With Trunk
Seizure
Enable Dial Tone Detection
(Outgoing)
Enable Answer Debounce
(Incoming)
Loop Break Duration (ms)
Tone Disconnect
Receiver/Transmission Gain
Trunk to Trunk Gain
Last Diagnosis Time
Diagnose button
Noise Level
Echo Return Loss
Description
Shows the result obtained the last time the Match Impedance button was clicked for that trunk.
Check to disable automatic impedance matching during system startup.
Select as None, FSK or DTMF for receiving caller ID digits. For North
America, the caller ID is FSK signal on analog trunk.
Specifies the Flash Duration time in milliseconds, with a range from 150 ms to 1000 ms.
When checked, if the trunk is set to Out of Service, the system will busy out the trunk. The CO will treat this trunk as a busy line and WILL NOT place a call to this trunk. (By default, this option is unchecked.)
When enabled, the trunk channel must detect outgoing dial tone prior to making the call.
Enables a timeout period of 2 seconds (for ignoring false CO disconnect signal), after answering an incoming call.
Disconnects signal if CO breaks loop current. You can set the duration from 200 to 1000 ms. 600 ms is common in North America.
Busy tone (reorder tone, fast busy tone, error tone, and so on) or dial tone (continuous tone, and so on). This should be used in conjunction with drop in loop current. For COs who cannot guarantee loop break, this may be the only option.
Slide setting adjusts the gain from -6 dB to 6 dB for every Triton Analog
Trunk channel.
The gain is not adjustable, by default. The user needs to run the diagnosis first to change the gain. The diagnosis process determines the max gain based on the diagnosis results.
The default setting is
0 dB
, and it is highly recommended that you not change this setting.
Caution! Setting the volume too high will cause distortion in voice quality and/or missed DTMF digits.
This configuration is to set Gain for calls that involve two analog trunks
(one in and one out). Because an analog trunk typically has energy loss of 3-12 dB, a two-trunk operation, like VM out call and MobileExt, may have low volume issues because energy loss is doubled. This configuration can compensate for the energy loss. The valid range is 0 to 6 dB.
Recommended value is 3 dB.
Caution: Setting the Gain too high may cause distortion in voice quality and DTMF tone. Your CO may not be able to recognize the dialing number if DTMF tones are distorted.
The last time the Diagnosis button was clicked.
Use this button to view the Noise Level, Echo Return Loss, and Hybrid
Echo Return Loss, measured using the current Impedance setting.
The noise level (displayed after you click the Diagnose button or the
Match Impedance button). Acceptable range for Noise Level is less than -67 dBm in value. For example, Noise Level of -72 dBm is good and -63 dBm is poor. You may experience high background noise and low voice volume if Noise Level is poor.
The measurement for echo return loss (displayed after you click the
Diagnose button or the Match Impedance button). Acceptable range for Echo Return Loss is less than -12 dB. For example, Echo Return
Loss of -19 dB is good and -8dB is poor. The IP phone users may hear their voice coming back (echo) if Echo Return Loss is poor.
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Triton Parameter
Hybrid Echo Return Loss
Rx Level at 600 Ohms
Test Rx Level button
Description
The measurement for hybrid echo return loss (displayed after you click the Diagnose button or the Match Impedance button). Acceptable range for Hybrid Echo Return Loss is less than -6 dB.
The Rx Level measurement at 600 Ohms, obtained by clicking the Test
Rx Level button. See Test Rx Level button, below.
Tests the receiving level of the trunk channel on a call to your local
CO’s Milli-Watt Test Number after you set the Impedance parameter to 600 Ohms and the Rx Gain to 0dB. Results are displayed in the Rx
Level at 600 Ohms field.
Performing Impedance Match on Your Own
For each individual analog trunk that is connected to the CO when the system starts up, MAXCS automatically selects an impedance profile to best match the Triton trunk interface with the CO. In the unlikely event that this automatic selection does not yield the optimal voice quality, you may want to disable the feature and select the best impedance by trial and error method.
To disable automatic impedance matching, check the Disable Impedance Match During System Startup check box.
Using the Match Impedance Button
Whenever a new analog trunk is connected to an empty port or is replacing an existing trunk, you will need to use the Match Impedance button to select the best impedance profile.
To do this, follow these steps:
1.
Click the Impedance Match button. While the impedance match is in process, you’ll see a “progress” box.
When the process is complete, the Match Impedance dialog box opens, with information relevant to this trunk:
Figure 97. Match Impedance dialog box
The Impedance parameter setting in the main dialog box is changed to the best match selection, and the measurement for noise and returned echo is performed with this impedance setting. The results of this measurement are displayed in the Diagnosis section of the main dialog box. The Hybrid Echo Return
Loss field shows the measurement before adaptation of the selected Impedance profile, and the Echo
Return Loss field shows the measurement after adaptation of the selected Impedance profile.
Note: If the Hybrid Echo Return Loss reading of a trunk is worse than -6 dB, for example, -5 db, the trunk may be subject to VoIP voice quality problems. Use this trunk to connect to analog phones only, or configure it to be the least used trunk. (Acceptable range for
Hybrid Echo Return Loss is -6 dB to -26 dB.)
Noise Level should be less than -67 dBm (acceptable range is -67 dBm to -90 dBm).
2.
Make calls from the trunks to test voice quality.
3.
Repeat steps 1 and 2 for all other trunk channels.
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If the Hybrid Echo Return Loss and Noise Level are not within the acceptable range, take the following steps to troubleshoot:
1.
Change the trunk to a different port on the Triton board, then diagnose again (this is to rule out a hardware problem).
2.
Check to see if any wire taps to the trunk wire (bridge tap). If so, remove them, then test again.
3.
Request the CO to check the trunk conditions, including Line Loss, and longitudinal balance.
The Match Result Button
Clicking the Match Result button shows you the result you got the last time you clicked the Match Impedance button for that trunk.
Figure 98. Impedance Match Result dialog box
Measuring the Rx Level of a Trunk Channel
In order to perform this test, you need to obtain the local CO’s Milli-Watt Test Number from your CO. When dialing this number, a 0dB tone is sent. For example, if your number is 510-252-9712, the Milli-Watt Test
Number from the local CO is 510-252-0020 (the prefix 510-252 is the same).
1.
Write down the Rx Gain, then set it to 0dB and click OK.
2.
Write down the Impedance setting, then change it to 600ohm, and click OK.
3.
Call the number you got from your CO, as noted above.
4.
Click the Test Rx Level button. When the test is complete, the Test Rx Level dialog box opens.
5.
Click OK. The Rx Level measurement is displayed in the Diagnosis section of the main dialog box.
If you call your local CO’s Milli-Watt Test Number, the acceptable range for Rx Level should be between
-6dB and -3 dB, with -5dB being ideal.
6.
Restore the Impedance and Rx Gain settings, and click OK.
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If You Need to Improve the Rx Level
If the Rx Level measurement is between -6 to -9 dB, and IP phones are used, take the following steps to increase the gain for the Triton analog trunk to IP phone connection:
1.
Go to VoIP Board configuration and click the Advance button.
2.
Increase the Transmitting gain to IP Extension to 9 for the Triton Analog Trunk. (Do NOT change the gain in the trunk property of the Triton Analog Trunk Board, since it may impact the echo canceller performance.)
If the Rx Level measurement is worse than -9dB (for example, -10 dB) you should contact the CO to adjust the line loss to the acceptable range.
If You Don’t Have the Milli-Watt Test Number
If you don’t have the local CO’s Milli-Watt Test Number, you can follow the steps below to measure the line loss when calling two local trunks:
1.
Copy C:\Post Office\Phrases\Lang1\phrase9900 to C:\Post Office\Phrases\LangCustom folder. Rename it an unused phrase name, for example, phrase0990 (the number must be less than 1000).
This phrase is a 1 kHz test tone.
2.
Select an unused AA and set the AA to play the prompt phrase you named in step 1 (0990 in this example).
Figure 99. Setting the AA to play a prompt phrase
3.
Set the Timeout to Repeat Current Level.
Figure 100. Setting Timeout to Repeat Current Level
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4.
Select a trunk as a testing reference – an analog trunk with a specific phone number is best – and set the trunk In Call Routing to the Test Line Loss AA.
Figure 101. Setting trunk In Call Routing to an AA
5.
Call from one trunk to the testing reference trunk. You should hear a 1 kHz tone playing at the originating side.
6.
While the tone is playing, measure the Rx Level at the trunk that is making the outgoing call.
If the reading is less than -6 dB, for example -3 dB, take the following steps to attenuate the gain for the
Triton Analog Trunk to IP phone connection: a.
Go to VoIP Board configuration and click the Advance button.
b.
Set the Transmitting gain to IP Extension to 3 for the Triton Analog Trunk. (Do NOT change the gain in the trunk property of the Triton Analog Trunk Board, since it may impact the echo canceller performance. If the reading is -6 dB to -14 dB, for example, -12 dB, no change is needed.
If the reading is -15 dB to -18 dB, take the following steps to increase the gain for the Triton Analog Trunk to IP phone connection: a.
Go to VoIP Board configuration and click the Advance button.
b.
Set the Transmitting gain to IP Extension to 9 for the Triton Analog Trunk. (Do NOT change the gain in the trunk property of the Triton Analog Trunk Board, since it may impact the echo canceller performance.)
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Incoming Call Routing
If the reading is worse than -18 dB, you should contact your CO to adjust the line loss to the acceptable range.
Incoming Call Routing
To set incoming call routing for a trunk, select the trunk on the General tab, then click the In Call Routing tab in the Trunk Configuration window. The trunk location shows in the title bar.
Figure 102. Trunk Configuration, In Call Routing tab
Regular Trunk Calls
For each trunk - or using “Apply to” to apply the settings to multiple trunks - you can set routing for the three
•
During Business Hours
•
Outside Business Hours
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•
Non Workdays
Within each of these three time slots, you have the following routing options for incoming calls:
•
Route to an extension selected in the list
•
Route to an auto attendant number selected in the list
•
•
Route to the operator
Web IP Calls
For web IP calls, you can set routing for the three time periods defined in the System Configuration window,
Business Hours tab (“Setting Business Hours” on page 56):
•
During Business Hours
•
Outside Business Hours
•
Non Workdays
Within each of these three time slots, you have the following routing options for incoming calls:
•
Route to an extension selected in the list
•
Route to an auto attendant number selected in the list
•
Route to the operator
Outgoing Call Blocking
To set outgoing call blocking for a trunk, select the trunk in the General tab, then click the Out Call Blocking tab in the Trunk Configuration window.
Figure 103. Trunk Configuration, Out Call Blocking tab
If you select Trunk allowed for Outside Calls at Any Time, call restrictions set in System Configuration,
Outcall Routing, and Extension Configuration still apply to calls made on the trunk.
If you select Outside Calls Allowed According to The Following Schedules, you can then use the Schedule
1, 2, and 3 options to set up to three different time periods during which calls are allowed. You can use Apply
to to apply the settings to multiple trunks.
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In Call Routing Configuration
In Call Routing rules determine how the system routes incoming trunk calls to various targets. The system’s routing steps are as follows:
Step
1
2
3
4
Routing Process
Match DID number configured in extension, workgroup, or hunt group. If there is no match, go to the next step.
Match caller ID defined in the Caller ID Routing table. If there is a match and
• today is a holiday, route the call according to the Holiday Profile’s routing rules.
• today is not a holiday, route the call according to business hour routing rules defined in the
Caller ID Routing configuration.
If there is no caller ID match, go to the next step.
Match DNIS number defined in the DNIS Routing table. If there is a match and
• today is a holiday, route the call according to the Holiday Profile’s routing rules.
• today is not a holiday, route the call according to business hour routing rules defined in the
DNIS Routing configuration.
If there is no DNIS number match, go to the next step.
If today is a holiday, route the call according to the Holiday Profile configured for the trunk port that the call is coming in on. If today is not a holiday, route the call according to the business hours routing rules defined in the In Call Routing tab of the Trunk Configuration window.
The In Call Routing Configuration window lets you enter Caller ID and DNIS numbers into a routing table and set routing rules for a matched number.
To configure In Call Routing, select PBX > In Call Routing Configuration.
Caller ID Routing
When an incoming call comes through a trunk with Caller ID, the system can route the call to the proper extension, to the auto attendant, or to the operator, based on the Caller ID number collected.
In order to locate an entry in the Caller ID table for an incoming call, a full match is required.
To access Caller ID routing, click the Caller ID Routing tab in the In Call Routing Configuration window.
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Figure 104. In Call Routing window, Caller ID Routing tab
Adding and Deleting Caller ID Route Entries
To add entries to the Caller ID routing table, click the Add button. In the dialog box, type a Caller ID Number and a descriptive Caller ID Name, then click OK.
The number and name entries have the following requirements:
•
The Caller ID Number field allows only 0-9, “-” (hyphen), and “*” (asterisk). For example, both
5102529712 and 510-252-9712 are acceptable.
•
The Caller ID Name is descriptive and optional; it can be used to remind you about the nature of the number and routing. For example, you might give the 2529712 number the name “Tech Support.”
To delete an entry, select it in the Caller ID number list, then click Delete.
Defining Caller ID Routing
After adding an entry, you define it by first selecting it in the list. When you select an entry, its name and other defined attributes, if any, appear in the fields of the tab. You can edit any of these attributes.
For each number, you can set routing for three distinct time periods defined in the Business Hours tab (see
“Setting Business Hours” on page 56):
•
During Business Hours
•
Outside Business Hours
•
Non Workdays
Within each of these three time slots, you have the following routing options for incoming calls:
•
Route to a particular extension selected in the list
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•
Route to a particular auto attendant selected in the list
•
Route to the operator
•
Reject call
Also, you can set additional routing attributes based on:
•
Holiday Profile – Routes incoming calls based on Holiday Profiles configured in System Configuration
(see “Routing Calls on Holidays” on page 58)
•
Business Hours Profile – Routes incoming calls based on Business Hours Profiles configured in System
Configuration (see “Setting Business Hours” on page 56). During Business Hours, Outside Business
Hours and Non Working Day are defined and selected by Business Hours profile.
•
Set Call Priority – Lets you assign a call priority from 1-9 to the selected caller ID number. The highest priority is 1, the lowest priority is 9.
•
Set Call SKLR – For workgroup-directed calls. Lets you assign a skill level requirement from 1-9 to the selected caller ID number. This setting tells the system to match the call to an agent’s skill level setting.
(Setting an agent’s skill level is explained in “Skill Based Routing” on page 268.)
•
Language Setting – Lets you specify that callers who dialed from the selected caller ID will hear prompts in the language you set here. This field will have choices only if you added sets of prompts according to
the instructions in “Multilingual Configuration” on page 89.
DNIS Routing
When an incoming call comes through a trunk with DNIS or DID numbers, the system can route the call to the proper extension, auto attendant or operator based on the DNIS or DID number collected.
In order to locate an entry in the DNIS table for an incoming call, a full match is required.
Note: For AltiGen SIP trunks, use 11-digit numbers when configuring DNIS routing.
To access DNIS routing settings, click the DNIS Routing tab in the In Call Routing Configuration window.
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Figure 105. In Call Routing window, DNIS Routing tab
Adding and Deleting DNIS Route Entries
To add entries to the DNIS routing table, click the Add button. In the dialog box, type in a DNIS Number and a descriptive DNIS Name, then click OK.
The number and name entries have the following requirements:
•
The DNIS Number must be the numbers 0 - 9 (the hyphen is not accepted in this dialog box). For example,
2529876 is an acceptable entry, but 252-9876 is not.
•
The DNIS Name is descriptive and optional; it can be used to remind you about the nature of the number and routing. For example, you might give the 2529876 number the name “Tech Support.”
To delete an entry, select it in the DNIS number list, then click Delete.
Defining DNIS Routing
After adding an entry, you define it by first selecting it in the list. When you select an entry, its name and other defined attributes, if any, appear in the fields of the tab. You can edit any of these attributes.
For each number, you can set routing for three distinct time periods defined in the Business Hours tab (see
“Setting Business Hours” on page 56):
•
During Business Hours
•
Outside Business Hours
•
Non Workdays
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DNIS Routing
Within each of these three time slots, you have the following routing options for incoming calls:
•
Route to a particular extension selected in the list
•
Route to a particular auto attendant selected in the list
•
Route to the operator
Also, you can set additional routing attributes based on:
•
Holiday Profile – Routes incoming calls based on Holiday Profiles configured in the System Configuration
window (see “Routing Calls on Holidays” on page 58)
•
Business Hours Profile – Routes incoming calls based on Business Hours Profiles configured in the
System Configuration window (see “Setting Business Hours” on page 56). During Business Hours,
Outside Business Hours and Non Working Day are defined and selected by the Business Hours profile.
•
Enable Fax-over-IP – Lets you enable the FoIP feature. (Note that FoIP support is best effort and can be dependent on the fax device make/model – MAXCS fax device configuration guides can be found in the
AltiGen Knowledge Base)
•
Set Call Priority – Lets you assign a call priority from 1-9 to the selected DNIS number. The highest priority is 1, the lowest priority is 9.
•
Set Call SKLR – For workgroup-directed calls. Lets you assign a skill level requirement from 1-9 to the selected DNIS number. This setting tells the system to match the call to an agent’s skill level setting. (Set-
ting an agent’s skill level is explained in “Skill Based Routing” on page 268.)
•
Language Setting – Lets you specify that callers who dialed the selected number will hear prompts in the language you set here. This field will have choices only if you added sets of prompts according to the
instructions in “Multilingual Configuration” on page 89.
FoIP In-Call Routing
For instructions on FoIP In Call Routing, see Fax-over-IP Configuration.
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Out Call Routing Configuration
There are two ways to initiate outbound dialing in an AltiGen PBX:
•
Using the trunk access code. The trunk access code is easy to configure and use. However, it does not have the capability to resolve complicated dialing situations.
•
Using the route access code. Using the route access code with the Out Call Routing table can resolve the following complicated dialing situations:
•
Multiple 10-digit dialing area codes.
•
Both 10-digit and 11-digit dialing in the same area code.
•
Multiple carriers providing trunks for different purposes. For example, you may have a local carrier provide trunks for local calls only and a long distance carrier provide trunks that can accept only long distance dialing.
•
Block certain dialing patterns by creating an exceptions list.
•
Assist VoIP hop-off dialing to another system.
•
Assist T1/PRI tie trunk hop-off to other system.
•
Assist system Zoomerang and client application dialing, for example, MaxCommunicator and
MaxAgent. For example, dialing from MaxCommunicator will carry 11 digits and require the system to remove a digit before making a call to the carrier if it is a 10-digit dialing area.
•
Divide trunks with the same characteristics into multiple routes and prioritize them when assigning routes on the Default Routes tab or on the Dialing Pattern tab of the Out Call Routing Configuration window.
When a user dials an outside number using the route access code, the system performs the following tasks:
•
Compares the dialed number with entries in the Dialing Pattern table. If there is a match, the system uses the route assigned to the dialing pattern to make the outbound call. The route assigned to the special dialing pattern may have a digit manipulation rule to add or remove digits from the dialed number.
•
If there is no match in the Dialing Pattern table, the system examines the digits to determine if the call is a local, long distance, international, or emergency call. The routes defined in the Default Routes tab are used to process the call.
Configuring Out Call Routing
To configure out call routing, select PBX > Out Call Routing Configuration.
The following configuration steps may help you configure out call routing correctly.
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1.
Before you configure Out Call Routing, make sure a route access code is configured in the System
Configuration window, Number Plan tab. If you have a problem changing a first-digit assignment in the
Number Plan tab to a route access code, you may need to set the Access Code in the Trunk
Configuration window for all trunks to None.
2.
Create a route and assign trunks to the route. Typically, different types of trunks will be grouped to different routes. For example, you may need to create a local route for local trunks, a long distance route for long distance trunks, and a VoIP route for IP trunks.
3.
Assign routes as Default Routes so that regular 7-digit, 11-digit, international, and emergency calls will go through.
4.
Solve a complicated dialing situation by adding an entry into the Dialing Pattern table and assigning a route to the specific dialing pattern.
5.
If the dialing pattern requires adding or removing digits, you may need to edit the Digit Manipulation on the Route Definition tab to solve the problem. Repeat steps 4 and 5 until all complicated dialing patterns are entered and configured properly.
6.
If a dialing pattern will use another system's trunk to hop-off, you may need to create a VoIP or T1/PRI tie trunk route and configure digit manipulation to indicate which system to hop-off to and how to tell another system that this is a hop-off dialing by adding a trunk access code or route access code in the dialing stream.
7.
If you would like to block a specific dialing pattern, add the dialing pattern and check Disallow this dialing
pattern check box.
8.
Fax trunk channels used for fax routing should not be added to Out Call Routing.
Failure to do so may cause failure of direct 911 dialing. If you do not want a user to call 911 directly because of too many 911 dialing errors, you can leave the 911 route not configured. In this case, you need to let all extension users know that they need to dial 9+911 to call emergency service. A proper warning sticker on the phone to notify employees about 9+911 dialing would be a good practice.
Some configuration examples are provided at the end of the chapter. Please use them as a reference to help you configure your dialing pattern correctly.
About Route Definitions
A route definition consists of a route name and group of trunks, listed in the order that the system will use for outgoing calls.
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Configuring Out Call Routing
Figure 106. Out Call Routing Configuration, Route Definition tab
Route Parameter
Route Index
Route Name
Digit Manipulation
Member Trunks
Not Member
Description
For identification purposes only.
Description of the route (maximum 30 characters).
You can insert or delete digits from the dialed number. See configuration samples to learn how to use digit manipulation in different situations.
Insert to Head: Insert a string of digits in front of the dialed number.
Delete from Head: Remove a string of digits from the beginning of the dialed number.
Displays the trunks assigned to the selected route. The order in which member trunks are added determines the order in which the trunks are used by the system when making an outbound call (the first trunk listed is used first, and so forth).
Displays all trunks that are not assigned to the selected route.
Creating a Route
1.
Click Add under the route definition list.
2.
Type in a name and index number, and click OK.
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3.
To add trunks to the route, select trunks from the Not Member list and use the selected trunks to the Member Trunks list.
button to move
4.
Use the Up and Down buttons to change the position of a trunk in the Member Trunks list. This is the order in which trunks are accessed.
5.
Click Apply.
Deleting a Route
Select the route you want to delete, and click the Delete button.
Setting Default Routes
You can set default routes for four types of outgoing calls: local, long distance, international, and
emergency.
Warning! It is important that you set up default routes right after routes are defined. Failing to do so will cause outbound dialing failure.
Click the Default Routes tab in the Out Call Routing Configuration window to configure default routes.
For each type of call, the system will use trunks specified in the
“1” field, if available, otherwise use trunks in the “2” field, and so on.
Figure 107. Out Call Routing Configuration, Default Routes tab
The above configuration means:
•
The system has a group of analog trunks and a T1 digital trunk from a local carrier that can accept local and emergency calls.
•
The system has a T1 digital trunk from a long distance carrier that can only accept long distance calls.
•
The administrator segmented local trunks into two routes, "Local Analog" and "Local T1". A "Long
Distance T1" route is created for the T1 from the long distance carrier.
•
When a user makes a local call, the administrator wants the system to use local T1 trunks first. If local T1 trunks are busy, then the system uses local analog trunks.
•
When a user makes an emergency call, the administrator wants the system to dial out from local analog trunks first. If local analog trunks are busy, the system uses the local T1 trunk.
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About Dialing Patterns
If your system is using a route access code, most likely you have one of the following situations:
•
Your area may have multiple 10-digit dialing area codes.
•
Your area may have both 10-digit and 1+10 digit dialing in a same area code.
•
Your system needs to borrow another system's trunk to make an outbound call over an IP or tie trunk.
•
You would like to block a dialing pattern in addition to system restriction setting.
Dialing patterns are exceptions. If you can, minimize the number of dialing pattern entries. Most companies don’t need to create dialing patterns.
Creating Dialing Patterns
1.
Click the Dialing Pattern tab on the Out Call Routing Configuration window.
Figure 108. Out Call Routing Configuration, Dialing Pattern tab
2.
Click the Add button.
3.
Type in the prefix and pattern length, and click OK.
4.
Assign routes to this prefix by selecting routes from the lists in the Route Priority section of the Dialing
Pattern tab.
5.
If this is a restricted number or pattern, skip step 4 and check the Disallow this dialing pattern check box.
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Deleting Dialing Patterns
Select the pattern you want to delete, and click the Delete button.
Dialing Pattern Configuration Tips
•
If a dialing pattern has multiple routes assigned to it, the system will try to use the first route configured to process the call that has this dialing pattern. If the first route is busy or not in service, the system will use the second route, and so on.
•
If a dialing pattern requires the system to add or remove digits, a route with digit manipulation configuration needs to be set up correctly. This means that you may need to have the same group of trunks belong to different routes. Each route may have a different digit manipulation rule.
•
If you are using dialing pattern to restrict outgoing calls, you need to be aware of the following system implementations:
•
The system first checks to see if the number is blocked for this extension (a setting in the Extension
Configuration window, Restriction tab).
•
The system then checks the System Configuration Call Restriction tab settings to see if this number is blocked by the system.
•
The system then checks the Dialing Pattern configuration, and if a specific number or pattern is not blocked, the system will dial the number through a proper route.
In other words, if extension and system call restrictions are not blocking a number or pattern, you can use
Out Call Routing to build restriction rules to block numbers or patterns.
Configuration Example – Solving 10-digit Dialing
Situation: Company ABC located in Dallas, area code 214, has one PRI circuit from the local carrier. Both 214 and 972 area codes are local 10-digit dialing area codes. The carrier will reject the call if the system dials 1214 or 1972 when dialing a local call.
Configuration Steps:
1.
Create a route to include all the T1 channels.
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2.
Apply the route to Default Routes.
Configuration Example – Solving 10-digit Dialing
3.
On the Dialing Pattern tab, add two dialing patterns: “1214” and “1972”, each with a pattern length of 11.
4.
Define a route called “10-digit Dialing” and add all T1 channels to the route. In the “Digit Manipulation” section, check the first box, select Delete from Head, and delete 1 digit:
5.
Apply the “10-digit Dialing” route to dialing pattern 1214 and 1972:
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Resolving Dialing Delay: Non-USA/Canada Countries
When installing the AltiGen system outside of North America, you may experience dialing delay when dialing through E1/PRI trunks that are using en-bloc (buffering digits and sending all digits at once). The system dialing logic may cause a 7-second inter-digit dialing delay for en-bloc trunks. To reduce the dialing delay, the following configuration is recommended:
1.
On the Number Plan tab in the System Configuration window, select a digit for route access.
2.
On the Route Definition tab of the Out Call Routing Configuration window, add a route definition entry for en-bloc and assign the member en-bloc trunk(s).
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3.
On the Dialing Pattern tab of the Out Call Routing Configuration window, add dialing pattern definition entries for the following prefixes:
• prefix = 0, length = 11
• prefix = 00, length = 14
• prefixes = 1-9, each length = 7
In the Route Priority field, use the list to select the En-Bloc route definition (assigned in step 2).
The Dialing Pattern tab should look as follows:
With this configuration, the system will see that all digits have been collected and will send digits to the CO, instead of waiting 7 seconds for the dialing to finish.
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Extension Configuration
The Extension Configuration window provides for creating extensions and setting their attributes. To open the
Extension Configuration window, either click the Extension Configuration button on the toolbar or select PBX
> Extension Configuration.
an IP extension, see “Setting Up IP Extensions” on page 209. To set up a mobile extension, see
“Mobile Extension Configuration” on page 229.
Figure 109. Extension Configuration window
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Chapter 14: Extension Configuration
There are three types of extensions:
•
Physical Extensions are associated with a physical port and device, usually a telephone set. This is what most users think of as an extension.
•
Virtual Extensions are not associated with a physical port. Virtual extensions can be used as message mailboxes and in telephone sharing environments. Users of a virtual extension can log in on any available station to access physical extension features using Feature Codes.
•
IP Extensions are generally associated with an IP phone. The option is unavailable when the Enable IP
Extension option is not checked. When Enable IP Extension is checked, it will allow the IP phone to log on as an IP extension.
About the Apply To Button
A change you make to an extension can often be applied to one or more other extensions by using the Apply
To button.
Clicking the Apply To button opens a list of all extensions to which the change can apply. Select the extensions to which you want to apply the change (all are selected, by default). Use the Shift or Ctrl keys to select several extensions.
The Apply To button is disabled unless a change you made can be applied to other extensions. When you use the button to apply changes to multiple extensions, it works on only those changed attributes that can be applied.
Setting up Extensions
Set up new extensions in the Extension Configuration window.
1.
Click the Add button below the Agent/Supervisor/Extension list.
Figure 110. The Extension parameters
2.
Type in an Extension Number.
The number must begin with a number assigned to be used for extensions, and it must be the length assigned to extensions, both of which are set on the Number Plan tab in the System Configuration win-
dow, as described in “Setting a System Number Plan” on page 52.
3.
If you have a multi-site setup, with multiple MAXCS systems connected over IP, a VoIP domain is created in the Enterprise configuration. If you want to publish the extension to all MAXCS systems within the VoIP domain, check the Global Extension check box. “(Global)” will be displayed beside the extension’s type in the Agent/Supervisor/Extension list. No configuration is needed on other MAXCS systems on behalf of this extension.
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Setting up Extensions
These are the benefits of making an extension a Global extension in a multi-site installation:
•
A user from any system only has to dial the Global Extension number, and MAXCS will resolve the routing through the VoIP domain setting.
•
Any user within the VoIP domain can forward voice mail to this Global extension.
•
The client applications MaxCommunicator and MaxAgent can see this Global extension number even it is not an extension in the local system.
4.
Select the Type of extension from the two options, Physical or Virtual. Unless this is an analog extension and you know the GatewayID/BoardID/Channel number, creating a new extension as a virtual extension is recommended. You can activate the extension from an analog or AltiGen IP phone by using
#27+password to log in. The system will determine the Gateway ID, Board ID, channel number, or IP address automatically.
5.
Depending on the type of extension you’re creating, take one of the following actions:
•
If you’re setting a virtual number, you’re done. Click OK.
•
If you’re setting up a physical extension, select an available physical location—gateway, board and
channel
for the line—then click OK.
The board ID and the channels (the ports) are displayed and available if they have not yet been assigned to an extension. Use the Next and Prev buttons in the Location section to select a location.
After you create an extension, you can set basic attributes on the Extension Configuration General tab. These attributes are discussed below.
Setting Personal Information
The top section of the General tab is for Personal Information:
•
First Name and Last Name of the extension user, each with a maximum of 32 characters.
Note: Only alphanumerical characters (A-Z,a-z,1-9) are supported for extension/group first name and last name. Symbols (such as “#”, “*”, “/”, “-”) are blocked, so as not to conflict with Dial by Name
(#34) and other feature codes.
•
Password for the extension user. The default is the system default password set on the Number Plan tab in the System Configuration window.
A valid password must be 4 to 8 digits (numbers or letters A-Z) in length and cannot be the same as its extension number. Basic password patterns, such as repeated digits (1111), consecutive digits strings
(1234), or digits that match the extension (Ext. 101 using 1012, 9101, 10101, and so on) are not allowed.
The letters map to numbers as follows:
4
5
2
3
Numbers Letters
A, B, C, a, b, c
D, E, F, d, e, f
G, H, I, g, h, i
J, K, L, j, k, l
8
9
6
7
Numbers Letters
M, N, O, m, n, o
P, Q, R, S, p, q, r, s
T, U, V, t, u, v
W, X, Y, Z, w, x, y, z
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•
Department – In an Enterprise VoIP domain, departments can be defined and extensions can be assigned to a department by using Enterprise Manager. When this is done, the department is displayed here.
•
Description – Optional descriptive information such as cubicle number or job title.
•
DID Number – Each extension can be assigned a DID number. This number does not have a fixed length, but the length must be long enough (range 2 - 16) for the system to match the DID incoming call.
If you configure a 10-digit DID number and inbound digital trunks only receive 4 digits, the last 4 digits of the DID number configured will be matched.
•
Transmitted CID – Each extension number can be assigned a caller ID number. When an outgoing call is made by this extension through PRI or IP trunks, the caller ID number entered in this field will be transmitted to the receiving caller.
When an extension user makes an outbound call through a PRI trunk, the system will transmit the Caller
ID based on the following rules:
•
If the Transmitted CID is configured, the number will be sent.
•
If the Transmitted CID is not configured, the DID number will be sent if it is a valid 10-digit number.
•
If the DID number is not configured or not valid, the Area Code and Phone Number entered in the
Trunk Configuration window will be sent.
•
If the Area Code and Phone Number are not configured in the Trunk Configuration window, the
System Main Number in the System Configuration window will be sent.
Note: These rules may be overridden by your PRI CID configuration or the SIP Trunk Profile you’re using.
•
E911 CID – A number entered in this field will be transmitted as the caller ID for 911 calls made by this
extension. See “Location-Based E911 (for Roaming Users)” on page 307 for instructions on setting up
E911 Location IDs for IP phones.
Note: If a number is not entered in the E911 CID field, the Transmitted CID is transmitted as the caller
ID for 911 calls made by this extension.
•
Language – Sets the language the extension user will hear for voice mail and system prompts. If voice mail and system phrases have been translated into other languages and properly added to the
C:\PostOffice\Phrases directory, the languages will be selectable from the Language list. (See
“Multilingual Configuration” on page 89 for information on adding translated prompts to the MAXCS
system).
•
Feature Profile – Sets an extension feature profile that includes enabling or disabling of extension features. The feature profile must first be configured by the administrator on the Feature Profiles tab of System Configuration.
•
A feature profile assigned to an AltiGen IP phone should have #26 enabled.
•
Enable Dial-By-Name – Select this box to allow incoming callers to search the extension list by employee name for this extension.
•
Enable Intercom – Select this box to enable the intercom call feature for this extension. Pressing #93 allows the user to make an intercom call to another intercom-enabled extension.
Note: Intercom is available for extensions on Triton Analog Extension Boards and AltiGen IP Phone
Extensions.
•
Agent – Allows the extension to be added as a member of one or multiple hunt groups or workgroups.
“
(Agent)” will be displayed in the extension’s Type field, next to the extension type.
•
Release SIP Tie-Link Trunk – This option has been added to support various AudioCodes devices. For full AudioCodes configuration instructions, refer to the appropriate article for your AudioCodes model.
Third-party configuration guides are stored in the AltiGen Knowledgebase (https://know.altigen.com).
This option instructs the SIP Trunk provider to release both the inbound and the outbound legs of a transferred call once the transfer has been completed.
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For a discussion of the options Enable SIP REFER and Enable Centrex Transfer, which are part of the
Release SIP Tie-Link Trunk feature, see the section SIP Server General Parameters.
Note: Not all PBX or services providers support SIP Refer or release link tie.
Account Code
These settings determine how callers use any account codes you have established when making outgoing trunk calls.
Figure 111. Account Code options
For information on creating account/code associations, see “Creating Account Codes” on page 62.
•
Enable Forced Account Code – Forces the user to enter an account code.
•
Override Allowed – Prompts the user to enter an account code, or the user can press # to bypass the account code.
•
Account Code Validation – Forces the user to enter a valid account code.
•
For Long Distance Call Only – The system determines if an outgoing call starts with a long distance or international prefix. If it does, the call will require an account code.
•
Block Account Code Display – The account code table will not be displayed when the user tries to tag the account from MaxCommunicator and MaxAgent. This prevents the user from seeing account codes they do not need to see.
Call Recording Options
The system administrator can specify the following non-workgroup call recording options for an agent extension.
Figure 112. Call Recording options
Warning! Listening in to or recording a conversation without the consent of one or both parties may be a violation of local, state, and federal privacy laws. It is the responsibility of the users of this feature to assure they are in compliance with all applicable laws.
License Assignment
•
Concurrent Session – When this extension is in recording state, a recording license is consumed; otherwise, a recording license is not being consumed by this extension.
•
Dedicated Seat – Assigns this extension a recording license for its exclusive use. The license is consumed whether or not the extension is recording.
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Recording Options for Non-Workgroup Calls
•
Disable – No recording of non-workgroup calls.
•
Auto record to central location – Records all non-workgroup calls, which are saved to a centralized lo-
cation (defined in System > Recording Configuration – see Configuring Call Recording on page 98);
this option requires either a shared Concurrent Recording Session license or a Dedicated Recording Seat license to be available.
•
Record on demand to central location – Records non-workgroup calls on demand, which are saved to
Recording Seat license to be available.
•
Record on demand to extension VM – Records non-workgroup calls on demand, which are saved to the extension’s voicemail box. No license is required for this option. If the recording file size is larger than the mailbox size set for the extension, the recording file is discarded. The administrator should assign a large enough mailbox size to this extension. (The mailbox size setting is on the Mail Management tab.)
Note: The recorded file will not be forwarded to e-mail as an attachment even if mail forwarding is enabled to forward voice mail to e-mail.
•
Record X out of 10 calls – If recording to a central location, automatically records all incoming non-
workgroup
calls at a specified interval for every 10 calls. Group calls are not recorded.
For example, if you set to record 4 out of 10 calls, the 1st-4th and 11th-14th, and so on, will be recorded.
The shaded calls will be recorded in the following example:
Recording Tone
•
Disable – No tone is played during a recording.
•
Insert tone before recording – Plays one recording beep to alert the parties that the conversation is being recorded.
•
Insert repeating recording tone – Plays a low-volume background beep every 15 seconds to alert the parties that the conversation is being recorded. The tone is recorded together with the conversation. The beep does not disrupt the conversation.
In MAXCS systems, the repeating tone is only available if one channels in use is SIP. With two TDM channels, repeating tones will not be available.
Note:
•
The recording session starts when the call enters the connected state and ends when hang up or flash is pressed, or when the call is transferred.
•
Except for license assignment, the recording setting in Extension Configuration only applies to non-
workgroup
calls. The recording setting in Workgroup Configuration only applies to workgroup calls. To allow an agent to record all calls (non-workgroup and workgroup), both recording settings must be enabled.
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Setting up Extensions
Physical Location and Type
You can change the extension’s type and location.
Figure 113. Extension Location and Type options
•
Type
The type of extension—physical or virtual—is set when you create the extension. After you create the extension, the type is displayed in brackets in the Agent/Supervisor/Extension list on the left side of the
Extension Configuration window.
You can change a Virtual extension to a Physical one, and vice versa.
If you change the type to physical, you can also set the location and configure the line as discussed in
the “Setting the Line Properties” on page 185.
For information about IP extension configuration, see “Setting Up IP Extensions” on page 209.
•
Location
•
Assigning a location to a physical extension – When changing a virtual extension to a physical extension, the Location parameters are available. If you know which board and channel this extension is wired to, you can use the Prev and Next buttons to select the correct board and channel number for this physical extension.
•
Changing the location – To change the location of a physical extension, select the extension number in the list of extensions, then click the Prev or Next buttons to change the board and channel settings until the location you want is displayed. Like other changes, this change isn’t finalized until you click
Apply.
Setting the Line Properties
For a physical extension, you can configure hardware options on the port used for the extensions. To do so, select the extension number in the list of extensions, then click the Line Properties button to open a dialog box that is specific to the board used for the extension.
Triton Analog Station Line Properties
If you select a Triton Analog Station Board extension and click the Line Properties button, you’ll see the Triton
Analog Station Line Properties dialog box.
You can also access this window by double-clicking a span in Channel Mapping List of the Triton Analog
Station Board configuration window.
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Figure 114. Triton Analog Station Line Properties dialog box
Configure the following hardware extension-specific features:
Triton Analog Station
Line Parameter
Caller ID Signal Format
Message Waiting Signal
Type
Flash Duration
Ringing Frequency (Hz)
Line Disconnect Signal
Description
Message format with which to send Caller ID information:
•
None
•
SDMF – Single Data Message Format for supporting and sending a single data type, such as phone numbers.
•
MDMF – Multiple Data Message Format for supporting and sending multiple data types, such as name and number information. (Default for
US/Canada installation.)
•
DTMF – Dual Tone Multi-Frequency, composed of high and low frequencies, for touch tone dialing.
Type of Message Waiting indicator for the phone set:
•
None
•
FSK/SDMF – Frequency Shift Keying/Single Data Message Format indicator.
•
FSK/MDMF – Frequency Shift Keying/Multiple Data Message Format indicator. (Default for US/Canada installation.)
Specifies the Flash Duration time in milliseconds:
•
85-750 (default)
•
50-600
•
100-700
•
150-800
•
200-900
•
300-1000
Select the frequency in Hz that is necessary for the equipment attached to this line: 28 (default) or 20.
The loop current break desired for answering supervision. Range 600-1000
ms (1000 ms is default).
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Triton Analog Station
Line Parameter
Caller ID Transmission
Methods
Receive from phone (dB
Gain)
Transmit to phone (dB Gain)
Description
Specifies how Caller ID will be detected:
•
CID between 1st and 2nd ring – Caller ID is received between first and second ring. (Most common in US/Canada)
•
DT-AS+CID prior to ringing – Dual Tone Alerting Signal Caller ID is received prior to ringing.
•
RP-AS+CID prior to ringing – Ring Pulse Alerting Signal Caller ID is received prior to ringing.
Range -3 ~ +3 db
You can decrease or increase the extension phone's talk volume with this setting. Default is 0 dB.
Range -3 ~ +3db
You can decrease or increase the extension phone's receiving volume with this setting. The volume will be lower or higher for the extension user. Default is 0 dB.
IP Extension Configuration
General tab.
Phone Display Options
For analog and AltiGen IP phones, the administrator can select what information is to be displayed.
Figure 115. Phone Display options
Depending on the number of display lines on the LCD, the phone can be set up to show two lines of specific caller information on the display.
In the Phone Display field, use the Number Line and Name Line lists to select the caller information to display:
•
Caller Number
•
Caller Name
•
DNIS Number
•
DNIS Name
•
AA Data
•
User Data
Note: For most phones, the number line can only display a number. If the Number Line is set to Caller
Name, DNIS Name, User Data or AA Data, the phone may display “Unknown” on the number line.
Alti-IP 600, IP 705, and IP 805 Phone Display Notes
For the Alti-IP 600, IP 705, and IP 805 models, the Name Line displays caller information under the following conditions:
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•
If Name Line is set to Caller Name, it will display caller name. If there is no name information, the number will be displayed.
•
If Name Line is set to Caller Number, it will display the caller number. If there is no number information,
“Unknown” will be displayed.
•
If Name Line is set to DNIS Name, it will display DNIS name. If there is no name information, the DNIS number will be displayed.
•
If Name Line is set to DNIS Number, it will display the DNIS number. If there is no number information,
“Unknown” will be displayed.
Configuring Group Options for an Extension
In the Extension Configuration window, Group tab, you can see the groups to which an extension is assigned, and you can change those assignments. Hunt groups are created in the Huntgroup Configuration window (see
window (see “Establishing Workgroup Membership” on page 266). Group members are assigned in those
configuration windows, as well.
Once a group is established, use the Extension Configuration window, Group tab, to configure hunt group and workgroup options for an individual agent extension, such as how much wrap-up time to allow that individual agent after a workgroup call.
You can assign an extension to and remove an extension from a group in the Extension Configuration window too. To assign an extension to a workgroup, the extension must be designated as an Agent extension. This is done on the General tab of Extension Configuration (check the Agent check box). A hunt group member does not have to be designated as an Agent.
To configure group options for an individual extension,
1.
Select the extension number from the Agent/Supervisor/Extension list in the Extension Configuration window. The extension number and type appear in the title bar of the window.
2.
Click the Group tab. You see a list of groups the extension is a member of and a list of groups the extension is not a member of. If the extension is an agent, both workgroups and hunt groups are shown. If the extension is not an agent, only hunt groups are shown.
Figure 116. Extension Configuration window, Group tab
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Adding or Removing Group Assignments
Adding or Removing Group Assignments
You can assign an extension to a hunt group in the Huntgroup Configuration window and to a workgroup in the
Workgroup Configuration window. Conversely, you can assign a hunt group or a workgroup to an extension in the Extension Configuration window.
To assign a group to the selected physical or virtual extension,
1.
On the Group tab, click the group number in the Not Member list.
2.
Click the Add button to move it to the Member list.
Note: If a hunt group or workgroup is configured to Ring All Available Members, the maximum number
of members is 20. See “Setting Call Handling Options” on page 246 for details.
To remove a group assigned to a physical or virtual extension,
1.
Click the group number in the Member list.
2.
Click the Remove button. The group moves to the Not Member list.
Note: You can use Shift+click and Ctrl+click to select more than one group.
Setting Wrap-up Time
You can set the Wrap-up Time for the selected physical agent extension. This option doesn’t appear for a vir-
tual extension or a non-agent extension
. Wrap-up time is a system delay between the time an agent finishes a workgroup call and the time the next call is routed to the extension. It gives the agent time to finish up with notes, prepare for the next call, log out of the group, or click the “Not Ready” button in MaxAgent. You can set a wrap-up time of up to 29 minutes, 59 seconds.
To set the extension wrap-up time,
1.
Check the Allow Workgroup Wrap Up Time check box.
2.
Using the lists, select the minutes and seconds for the delay. Be sure to set at least enough time (for example, 5 seconds) to allow an agent to click the “Not Ready” button in MaxAgent after putting the caller on hold and going onhook.
Setting Inter Call Delay
This configuration applies only to calls waiting in queue. The Inter Call Delay can create a time delay before the next workgroup call in queue rings the extension after the extension finishes one of the following activities:
•
Makes an internal or outbound call
•
Receives a direct inbound call
•
Accesses voice mail
It is possible that an agent may execute one of the above activities during the wrap-up period after finishing a workgroup call. The following rules govern which delay timer will take effect:
•
If Wrap-up time is still active, the Inter call delay will be ignored.
•
If Wrap-up time is expired when one of the above activities is completed, the Inter Call Delay will be applied. The system will not pass a workgroup call to an agent until Inter Call Delay is expired.
To set the extension Inter Call Delay time,
1.
Check the Inter Call Delay check box.
2.
Using the lists, select the seconds for the delay.
Picking Up a Call from the Workgroup Queue
Check Allow pickup call from workgroup queue to allow a MaxAgent user to pick up a call from the workgroup the agent belongs to. The agent needs to be in the log-in state to be able to pick up a call from the queue.
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Logging Outbound Workgroup Calls
You can assign an agent to an outgoing workgroup, which is useful for call detail reporting and workgroup statistics. All calls made by the agent while logged into the workgroup will be tracked as calls from the workgroup.
The agent’s outgoing workgroup can be assigned to any workgroup of which he is a member.
To set an agent’s outgoing workgroup,
In the Log Outbound Call to Workgroup field, use the list to choose a workgroup from among the workgroups the agent belongs to. If the Allow agent to change check box is selected, the agent can change the outgoing workgroup from the phone set by using feature code #53 or from MaxAgent.
When a user is first assigned to a workgroup, it is set as their default outgoing workgroup and remains so no matter how many workgroups the user is subsequently assigned to. If an agent is unassigned from their outgoing workgroup, the outgoing workgroup is automatically set to N/A.
Setting up Station Speed Dialing
For each extension, you can set up to 20 station speed dial numbers. The numbers available are from 00–19, and are entered by the user following the extension speed dial access code, #77.
To work with Speed Dialing settings, click the Speed Dialing tab, then select the extension you want to set speed dialing for.
Figure 117. Extension Configuration, Speed Dialing tab
Editing Speed Dial Entries
To add or edit an entry,
1.
Double-click the Station Speed ID number you want to work with, or select the number and click Edit. Or click Add to add an entry.
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Setting Mailbox Options
Figure 118. Speed Dial details
2.
Select the ID number using the arrow, type in a name for the Speed Dial entry, then the full number as you would dial it, with a maximum of 20 digits per entry. For example, the phone number 914085551212 comprises 9 (trunk access code), 1 (long distance prefix), followed by 408 (area code), and finally the seven digit telephone number.
Valid digits include 0 through 9, #, *, and (,) comma. The comma represents a one-second pause.
Setting Mailbox Options
The Mail Management settings define how voice messages are handled for an extension: whether the mailbox is information only or is full-featured, how messages are announced and processed, and how much capacity is allotted to message storage.
To work with mailbox settings, select the extension number you want to work with from the Agent/Supervisor/
Extension list, then click the Mail Management tab.
Figure 119. Extension Configuration, Mail Management tab
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Setting an Information-Only Mailbox
You can check the Information Only Mailbox check box to set virtual or physical extension mailboxes to Information Only, then click Apply to to set one or more extension mailboxes.
An Information Only mailbox allows callers to listen to customized recorded announcements. To repeat the announcement, callers are instructed to press the # key. This mailbox does not take messages from the caller.
Disabling a Mailbox
When you disable a mailbox, a special greeting is played to announce that this mailbox is not accepting new messages.
Assign Exchange Integration License
Check this check box if the selected extension is to be integrated with Microsoft Exchange.
SMTP/POP3 Setting
•
Email Name – the user’s e-mail name without the @domain. The default e-mail name is ext[extension
number],
that is, the letters “ext” followed by the extension number. For example, the default e-mail name for extension 2497 would be ext2497.
•
Retrieve Voice Mail by Email Client – selected, this sends voice mail to the user’s e-mail as an attachment.
Mail Forwarding Options
•
Enable Mail Forwarding – selected, the user’s e-mail will be forwarded to the e-mail address you specify in the Forward Email Address box. The address should be a full address, including the domain (for example, [email protected]).
If you enable mail forwarding, you also specify what you want done with the original messages after they have been forwarded. In the drop down list you can choose to:
–
Delete Messages after Forward
–
Keep the Messages as New
–
Keep Messages as Saved
Setting Message Playback Options
You can use the following check boxes to turn on or off options for listening to playback of recorded messages.
These options apply to both new messages and saved messages, and they can be applied to multiple extensions using Apply to.
Message Playback
Parameter
Announce Message
Sender Before Playback
Announce Time Stamp
Before Playback
Confirm Callback
Number
Description
Selected, the user hears the type of the message sender (internal or outside) before listening to recorded messages.
Selected, the user hears the timestamp (time and date) of each message before playback.
Selected, the system reads back the caller’s number and asks the caller to confirm.
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Setting Message Notification Options
Message Playback
Parameter
Enable Distinctive Call
Waiting Tone
Play the Newest Voice
Message First
Description
Selected, the extension user will hear a "beep" tone when there is a call waiting in the extension's queue.
Selected, new voicemail will be retrieved first. When not selected, the system will play voicemail based on first-in-first- out (FIFO).
Press Zero Option
This option allows a caller to press “0” while listening to this extension’s greeting. Use the list to select one of the following forwarding destinations for the call: Voice Mail, AA, Extension, Group, Operator (default),
Outside Number, Application Extension, or Line Park. When the caller presses “0”, the call will forward to the specified destination.
Setting Mailbox Capacities
You can set various mailbox capacities with the following options:
Mailbox Capacity
Parameter
Max Number of
Messages
Mailbox Size
Max Message Length
Retention Length of
Saved Messages
Description
Maximum number of messages stored in the user’s mailbox. The range is 1–
999, defaulting to 100.
Mailbox size in MBs of stored messages. The range is 1–500 MB, with a default of 50.
Maximum length of voice messages in minutes. The range is 1–30 minutes, with a default of 5 minutes.
Number of days saved messages are archived by the system. The range is 1–
90 days, with a default of 60.
These options can be applied to multiple extensions using Apply to.
Setting Message Notification Options
The Notification tab of Extension Configuration provides for setting notification options on new incoming email as well as voice messages.
To work with notification settings, select the extension number from the Agent/Supervisor/Extension list, then click the Notification tab.
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Figure 120. Extension Configuration, Notification tab
Individual users can also configure Message Notification within the AltiGen client applications
MaxCommunicator and MaxAgent.
Apply To Button” on page 180 for more information on using Apply to.
Setting the Message Types for Notification
Select the types of messages for which the extension user is notified:
•
None – No notification. Selecting this option does not prevent the user from getting message waiting indicators or stutter dial tone when new messages are received.
•
Urgent Voice Messages Only
•
All Voice Messages
The system will perform notification under the following conditions:
•
Extension's message notification is set to Urgent Voice Messages Only.
•
Extension's notification Schedule is set to Non-Business Hours.
•
Voice mail received during business hours is marked urgent.
•
Extension user does not check the urgent message.
The system will start notification as soon as it enters non-business hours.
Note: Message notification can also be set in MaxCommunicator and MaxAgent, and the settings are reflected in MaxAdministrator.
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Setting Message Notification Options
Emergency Notification
When any extension dials an emergency number, the system can make calls to specified extensions, groups, or outside numbers. To configure this option, select the extension/group/outside number, and check the When
Emergency Number Has Been Dialed check box.
Unusual VM Activity Notification
When certain unusual activity is detected from an extension’s voice mail, the system can notify a designated extension. This option is intended to help detect if a hacker has obtained control of and is making calls from an extension’s voice mail. To alert an extension (usually the administrator) when either of the following abnormal activities are happening, select the extension and check the option When unusual call activity has been
detected:
•
When calls made from voice mail are unusually long (by default, more than 120 minutes)
•
When the number of calls made from voice mail is unusually high (by default, more than 20 calls in one voice mail session)
When the designated extension is notified, the system will play "Unusual call activity has been detected from
Extension xxx. More than yy calls have been made from the extension's voice mail. Please verify with the extension user." Or "Unusual call activity has been detected from Extension xxx. The extension made more than a yyy-minute call from the extension's voice mail. Please verify with the extension user." The security notification will be made only once within a call.
Setting Parameters for Unusual VM Activity
To change the parameters for the number of calls or length of a call, you must add the following strings and values to the Windows registry:
•
SecurityConnectionDuration
(value range is from 1-1440 minutes [24 hours]). When the setting is out of range, the default of 120 minutes will be used.
•
SecurityNumberOfCalls
(value range is from 1-100 calls). When the setting is out of range, the default of
20 calls will be used.
Adding security values to the registry
To add one or both of the above security values to the Windows registry:
1.
Choose Run from the Windows Start menu, type regedit, and click OK.
2.
Go to HKEY_LOCAL_MACHINE\SOFTWARE\AltiGen Communications, Inc.\AltiWare\InitInfo.
3.
On the right side of the Registry window, right-click and choose New > DWORD Value.
4.
Type one of the security strings listed above, then double-click the entry.
5.
Choose Decimal as the Base option.
6.
Type the value you want (see the allowed range listed above) in the Value data text box, and click OK.
7.
The value you enter appears in parentheses in the Data column.
8.
For the values you entered in the registry to take effect, from the MaxAdministrator menu, choose
Diagnostic > Trace. The Trace Filter dialog box opens. Click the Minute Task button in the dialog box.
Alternatively, you could restart the system for the values to take effect.
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Where Security Alerts Are Logged
Security alerts are logged to ...\AltiServ\Log\SecurityAlert.txt. The log includes date, time, extension number, pad number, and the alert reason. Emergency calls are also logged to this file. Following are some examples:
2007-02-04 08:30:25 Extension 212 made more than 20 calls from voicemail(1:2)
2007-02-04 16:00:50 Extension 395 made more than a 120-minute call from voicemail(0:6).
2007-02-18 09:05:32 Extension 395(2:3) made an emergency call-###.
Note: A SecurityAlert.txt file does not appear in the ...AltiServ\Log folder until a security alert event has created it.
Setting the Type of Notification
There are four options for sending the notification or reminder message: phone, pager, extension or custom
application (Custom App).
•
Extension – To use the Extension option, select the Extension radio button, then type the extension number into the text box.
•
Phone/Pager – For the Phone and Pager options, first specify the trunk or route access code using the list next to the Phone radio button. The Any option means to locate any available trunk. Then type in the number with all relevant dialing prefixes other than the trunk code, using a maximum of 63 digits.
•
Custom App – When used in conjunction with a third-party notification application, the Custom App feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log-on extension to which the third-party application is connected. Contact your local
AltiGen Partner for more information on using this feature.
Note: The Reminder Call will not work with this selection.
Note also the following considerations:
•
For the Pager option, the system calls the specified pager number and then dials the system main number
(as set in System Configuration, General tab), which is then displayed on the user’s pager.
For the operator-assisted paging function, the operator phone number and the pager number must be entered in the <phone number>*<pager number> format. For example, if the phone number to call the pager operator is 7654321 and the pager number to page the user is 12345678, the notification outcall number that needs to be entered is 7654321*12345678. When the pager operator answers the Message
Notification call, MAXCS announces the pager number and the System Main Number (as configured on the General tab of System Configuration), which will be displayed on the user’s pager. The operator is also given the option to repeat these numbers by pressing ‘#’.
Outcall to Cellular or PCS Phone Numbers
When an outcall is made by the system (for One Number Access, Message Notification, Zoomerang, Call Forwarding, and so on) to a cellular or PCS phone, it may ring the phone once but not necessarily present the call and make a connection. This will happen if the ringback tone played by the cellular service provider does not conform to standard ringback tones. To work around this problem, append a few commas (,) to the outcall (cellular) number when entering it. Each comma provides a one second pause.
Setting Notification Timing
When notification is configured to an outside phone number, the system will announce, "This is the outcall notification message for…" after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the carrier. If the system plays the announcement phrase before the notification call is answered, the phrase will be cut off. The following two options can be configured based on answer supervision capability:
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Configuring Calling Restrictions
•
Seconds after Dialing – If the carrier of the outside phone number cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
Note: If the delay is set too long, the notified party will hear silence before the announcement is played.
•
Seconds after Answered – This field is set to 0 seconds and it is not configurable for notification to a phone number. It means the system will play the announcement immediately after answer supervision is received.
When notification is configured to a pager, the system will transmit DTMF digits as the return phone number
(the System Main Number as set in the System Configuration General tab) after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the pager system. If the system sends digits before the call is connected, some digits will be cut off. The following two options can be configured based on answer supervision capability:
•
Seconds after Dialing – If the pager carrier cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
•
Seconds after Answered – If the answer supervision signal is provided by the carrier, check this option and set the delay timer to 2 to 5 seconds. In some cases, the pager carrier cannot detect DTMF right after the call connection. (Default is 10 seconds, maximum is 30.)
Note: You may need to try a different delay setting to make sure the user return number is transmitted properly after configuration.
Setting Notification Business Hours
You can choose one of three options for when the extension user is to be notified of new messages:
•
Non-Business Hours – Notify only during non-business hours. Business hours are set in System
Configuration, Business Hours tab (see “Setting Business Hours” on page 56).
•
From/To – Notify during a specified time of day. Select the hours in the From and To time scroll boxes.
•
Any Time – Notify at all times (every day).
Enabling Message Notification
After configuring your message notification settings, to enable message notification, check the Allow
Extension User to Configure Forwarding, Notification and Reminder Call to an Outside Number check box on the Restriction tab of Extension Configuration.
Configuring Calling Restrictions
To work with extension call restrictions, select the extension number you want to work with from the Agent/
Supervisor/Extension list, then click the Restriction tab.
Apply To Button” on page 180 for more information on using Apply to.
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Figure 121. Extension Configuration, Restriction tab
Setting Call Restriction Options
You can use one of the following options in setting restrictions on an extension or on multiple extensions using
Apply to.
•
No Restrictions on Outcalls
•
Internal Calls Only – extension-to-extension.
•
Internal, Local, and Unrestricted Area Codes – Allow extension to call internal, local, and area codes defined in the Unrestricted Area Codes in the Call Restriction tab of the System Configuration window.
•
Allow Internal/Local/Unrestricted, and Defined Prefixes – In addition to the above privilege, allow the extension to call prefixes you specify in the Prefixes Allowed boxes. Include all relevant prefix numbers
(for example, if appropriate, you would include 1+area code before the number). This configuration will not override System Prohibited Prefixes set in System Configuration.
•
All Calls Allowed Except the Defined Prefixes – In addition to System Prohibited Prefixes, you can block this extension from dialing the numbers defined in the Prefixes Disallowed boxes.
Setting Other Call Restrictions
Other call restriction rules can deny or allow the following:
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Setting Answering Options
•
Allow Calls to be Transferred or Conferenced to an Outside Number – when checked, the internal extension user can log into voice mail, make a call to a second party, then transfer or conference to a third party.
•
Allow User to Configure Forwarding, Notification, and Reminder Call to an Outside Number – This setting regulates extension call forwarding, voice mail notification, and reminder call configuration. If this setting is not checked, you will see a warning message pop up when trying to set up forwarding to an outside number. International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Return Calls from within VM System – when checked, an outside caller can dial into the system, log in to the extension’s voice mail, and make or return calls from the voice mail (Zoomerang feature). International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Forward International Calls from within VM system – This setting regulates making international calls from voice mail and forwarding to an international number. You need to check the second and third options to be able to check this configuration.
Caution!
Allowing any of these options may increase the potential for toll fraud. Make sure the password is properly configured to prevent an intruder from using this voice mail box to make an outbound call.
AltiGen recommends that you leave the fourth option unchecked for all extensions at all times.
Setting Answering Options
Answering options include forwarding, handling busy calls, handling no-answers and other options. Which options are available depends on the type of extension. Virtual and physical extensions each use somewhat different answering options.
the type of extension, you can only apply the choices to the same type of extension.
For example, If you are working with the settings for a virtual extension, you can use Apply to to apply changes to one, some, or all virtual extensions, but not to physical extensions.
To work with extension answering options, select the extension number from the Agent/Supervisor/
Extension list, then click the Answering tab.
Figure 122. Extension Configuration, Answering tab
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Forwarding All Calls
Call Forwarding is available to all types of extensions.
This is the Call Forwarding feature that is also accessible by the extension user by dialing #36.
One Hop Limit to Call Forwarding for a Transferred Call
There is a one hop limit to call forwarding when the call that is being passed is a transferred call. For example, extension 100 receives a transferred call and forwards this call to extension 101; extension 101 is set to forward all calls to extension 102; extension 102 receives the call but CANNOT forward this call to another extension.
10-Hop Limit to Call Forwarding for Direct Calls
For direct calls, there is a “10-hop” limit to call forwarding. For example, extension 100 forwards to extension
101, 101 forwards to 102, 102 forwards to 103, and so on, through extension 120. A call to extension 100 will be forwarded to 101, which will forward to 102, which will forward to 103, and so on, until the call has been forwarded 10 times. At this point, the call will not be forwarded again; if the last extension in the forwarding chain does not answer, the call is sent to extension 100’s voice mail.
If there is a loop condition in the forwarding chain (for example, 100 forwards to 101, 101 to 102, and 102 back to 100), the call is sent to the first destination’s voice mail.
To enable call forwarding, check the Enable Call Forward to check box, then, using the list, indicate the forwarding destination. You can use Apply to to act on multiple extensions, with the restrictions discussed in the previous section. The forwarding options are as follows:
•
To Voice Mail
•
To AA – Select the auto attendant number to use in the list under the option.
•
To an Extension – Select an extension from the list.
•
To a Group – Select a group from the list.
•
To the Operator
•
To an Outside Number – This option is available if it is allowed in the Other Call Restrictions option in
Cellular or PCS Phone Numbers” on page 196.
If you choose Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
•
To an App Ext – When used in conjunction with an SDK-based application.
•
•
To Free Format – This option is available only to virtual extensions. See the section “Free Format
•
To Paging Trunk – This option is available only to virtual extensions. To use this option, you have to select a paging trunk in Trunk Configuration.
Note: Forwarding calls to a pager is possible but not recommended since callers will only hear what is heard when calling a pager and will not know to enter a return phone number unless instructed.
Free Format Forwarding
You can configure an extension to send out additional DTMF digits to an extension, hunt group/workgroup, or out-side number after the call is forwarded to an outside line.
Release 8.0 supports forwarding for SIP Trunk and SIP extensions.
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Setting Answering Options
There are various uses for this virtual-forwarding feature. For example, you can configure an extension to forward fax calls to the first available fax hunt group. Another example would be to forward calls to a
FaxFinder extension. You can embed several commas to add a delay before MAXCS releases the
Centrex line to complete a call transfer.
Free format is supported on SIP trunks, PRI trunks, T1 trunks, analog trunks, SIP extensions, and analog extensions. Using an IP extension, APC extension, or Paging Group as a forward target is not supported.
Forwarding over IP and E1 trunks is not supported.
You can embed commas to insert a delay; each comma inserts a one-second delay. For a trunk call, the wait time starts right after the digits are dialed (even while the target phone is ringing). For an extension call, the wait time starts after connecting to the extension (it does not start when ringing begins).
Example: Forwarding to a FaxFinder Extension
Suppose that you want an extension to forward incoming calls to a FaxFinder extension 2002 which is behind an AudioCodes MP202 or MP118 device. This configuration is illustrated in the following figure.
Figure 123. Example of call forwarding with a 3-second delay and DTMF digits to send
The string in the figure above indicates that the call should be forwarded to extension 2002 (for FaxFinder).
The next three commas each insert a one-second delay, for a total delay of three seconds. The last three digits indicate to send DTMF digits 213.
After the DTMF digits are received, FaxFinder will use "213" as the destination number to receive the Fax.
Example: Forwarding to an Outside Number
Suppose you have a virtual extension 2001 and you want to set call forwarding to an outside number
"4085979000" through the SIP trunk.
Figure 124. Example of call forwarding to a specific extension at an external phone number
The string in the figure above indicates to forward the call to that outside phone number (our example is
AltiGen's corporate number) and then wait for 8 seconds (this is the 8 commas). After that delay, three more DTMF digits will be sent out through the SIP trunk. The result is that the call will be sent to extension
213.
Format Guidelines
•
Each comma inserts a one-second delay after the call is forwarded. We recommend that you use at least five commas (for five seconds). Longer call setup time may require additional commas. However, too many commas will impact the cut through time.
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•
You can enter up to 40 digits.
•
You can include the digits 0-9, *, #, and ",".
Configure Forwarding
To configure this forwarding,
1.
Select PBX > Extension Configuration.
2.
Select the extension and switch to the Answering tab.
3.
Check Enable Forward to and set it to Free Format.
4.
Enter the appropriate string in the next field.
Figure 125. Example of call forwarding to an outside phone number with a long delay
Do Not Disturb
Enable Do Not Disturb – Check this option to send all calls for the selected extension(s) to the extension’s voice mail. This feature is also accessible by the user at the user’s station by dialing #33. Note that this overrides any One Number Access settings for the extension. Polycom phone users can press the DND button if their extension has an Advanced Features license.
Handling Busy Calls
You have several options for handling calls while the extension is busy, and again, the options vary depending on the extension type. If you do not enable busy call handling, the caller simply hears a busy signal.
To enable the options, check the Enable Busy Call Handling check box, then select from the following options:
•
Forward to Extension – Select an extension number in the drop-down list. See “10-Hop Limit to Call
Forwarding for Direct Calls” on page 200.
•
Forward to Voice Mail
•
Place Caller in Queue – Places caller in the extension’s personal queue. This option is available only if
Multiple Call Waiting or Live Call Handling is turned on.
•
Forward to AA – Select the auto attendant number to use in the list under the option.
•
Forward to Line Park – Use the list to select a Line Park group to route the call. (See “Line Park
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Configuring One Number Access
Setting Call Waiting Options
Call waiting options are available only if the Enable Busy Call Handling check box has been checked.
•
Enable Single Call Waiting – Sets up single call waiting. This feature gives an alert tone (audio beep) to indicate that a call is waiting. This feature must be enabled in order to conference incoming calls.
•
Enable Multiple Call Waiting – Enables a “personal queue” of multiple calls waiting. This allows the user to transfer or park the current call before picking up the next call in queue.
•
Enable Live Call Handling – This feature is mainly for the system operator. It allows callers to stay in the personal queue while the extension user is checking voice mail or operating other features. The caller will hear a ring back tone while in queue. The call will be shown as "ringing" on AltiConsole.
Note: On a Polycom phone, the incoming call will follow the extension’s RNA handling rules instead of showing as “ringing.”
Handling Unanswered Calls
The No Answer Call Handling function provides options for handling calls when no one answers the extension within a specified number of rings.
Except for Enabling One Number Access, these options are not available to virtual extensions.
To enable these options, check the Enable No Answer Handling check box.
Use the Number of Rings Before Handling scroll box to select a number between 2 and 20 for the times the telephone rings before the call is handled by the system.
Select one of the following options for no answer call handling:
•
Forward to Extension – Select an extension number in the drop-down list. See “10-Hop Limit to Call
Forwarding for Direct Calls” on page 200.
•
Forward to Voice Mail
•
Forward to AA – Select the auto attendant number to use in the list under the option.
•
Enabling One Number Access
This check box option is available to all extension types, but with qualifications:
•
It is available to physical extensions only when the Forward to Voice Mail option is selected.
•
It is not available when Forward to AA, Forward to Extension, or Forward to Line Park is selected.
Configuring One Number Access
One Number Access (ONA) gives the caller an option to find the extension user when the extension is ring no answer. Caller still has the option to leave a voice mail if the system is unable to find the extension user.
Note: Options on the tab are disabled unless One Number Access has been enabled as a No Answer option on the Answering tab of the Extension Configuration window.
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Figure 126. Enable One Number Access option on the Answering tab
Also, if the Enable Do Not Disturb option is selected in the Answering tab, the call is forwarded to voice mail regardless of ONA settings.
To configure ONA, select the extension number from the Agent/Supervisor/Extension list, then click the One
Number Access tab.
Figure 127. Extension Configuration, One Number Access tab
One Number Access Options
In the One Number Access tab, use the list to select an option for One Number Access:
•
Disabled
•
Enabled at any time
•
Enabled during business hours only
•
Enabled during non-business hours
•
Enabled based on schedule
If you select this last option, Enabled based on schedule, you can then select and set up to four different time periods using the From and To time lists.
After choosing any of the enabling options, you set the Verify Caller ID and Forwarding choices, and these are discussed below.
Note: You can also enable and set up One Number Access remotely through MaxCommunicator.
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Configuring One Number Access
Disabling One Number Access
You can disable ONA for the extension by selecting the Disable option. Selecting Disable on this tab does not destroy the data you might have entered. For example, if you entered a group of Caller IDs to use to identify the caller, these will be available if you enable one number access at a future time.
Call Screening
When the Enable Call Screening option is checked, callers accessing One Number Access will be prompted to record a name in order to continue the ONA process. The recorded name is played after the callee (ONA target) answers the call and optionally enters a correct password. The callee will then hear the caller’s name and can decide whether or not to accept the call.
Setting Caller ID Verification
You can check the Verify Caller ID based on the following check box and then type in up to 10 phone numbers in the text boxes. Whenever the system detects a call from one of the numbers entered here during the selected schedule, the system searches for you by dialing the numbers configured in the Forwarding
Number fields.
Caution!
If ONA is enabled and no numbers are entered for Caller ID Verification, ONA is available to all callers.
Caller ID verification entries should be complete phone numbers.
Using a Password Verification
You can also enter a random “password” number such as “5555” so that any caller who knows this password can use ONA to find you, regardless of where they are calling from. Once you’ve set this up, you need to instruct the caller to dial 1 during your personal greeting, then enter the “password” to use ONA.
Specifying Forwarding Numbers
The Forwarding Numbers are used by the system to find the user when ONA is active. You can set up to four different numbers. When ONA is active, the system dials the forwarding number(s) in the order they are displayed on the One Number Access tab. The Forwarding Number order does not correspond to the
Schedule order.
You can forward to another extension, or to an outside number. You can use an outside number only if the extension is set to allow for Transferred/Conferenced/Forwarded calls on the an Extension Configuration
Restriction tab under Other Call Restrictions.
When you use the outside number option, select a trunk or route access code in the list and type in the phone number as it would be dialed after keying the access code.
Check the Check Password option to force users to enter their extension password when a call is forwarded to them via ONA. This ensures that only the owner of the extension can answer the call.
You can set the ONA ring duration from 5 to 45 seconds using the Ring for ... seconds list. Default value is
20 seconds. The system will ring the ONA target within the specified time limit. If the ONA call is not answered within the ring duration, the system will terminate the ONA call. This option will prevent a cell phone voice mail from answering the ONA call and recording the ONA announcement phrase into the cell phone voice mail box.
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Setting Up Monitor Lists
The Monitor List tab provides for setting up lists of extensions for which call processing events can be monitored by the extension user. Once a monitor list is established, the application logging into the extension can receive call events for the monitored extensions. The monitor list is available in the MaxCommunicator and
MaxAgent Monitor windows, AltiConsole, and in Line Monitoring events in AltiGen SDK.
Warning! Listening in to or recording a conversation without the consent of one or both parties may be a violation of local, state, and federal privacy laws. It is the responsibility of the users of this feature to assure they are in compliance with all applicable laws.
Restrictions and Defaults
•
Monitoring is effective for physical and virtual extensions; physical and virtual extensions have monitoring rights, and can be monitored. If you place a physical or virtual extension in a Monitor List, that extension will show in the client application’s Monitor window.
•
If you add an extension (1001, for example) that belongs to Workgroup A to the Monitor List for a member of Workgroup B, the Workgroup B member will only be able to pick up personal calls to 1001, not workgroup calls.
•
In MaxSupervisor, the user can monitor only the workgroup(s) he or she logs in to, regardless of the monitoring rights assigned to his or her extension in MaxAdministrator.
Configuring a Monitor List
To set up a monitor list, select the extension number to receive the monitoring rights from the Agent/
Supervisor/Extension list, then click the Monitor List tab.
To block the extension from seeing caller names and numbers, check this box
Figure 128. Extension Configuration, Monitor List tab
To block the extension from seeing any Caller Name and Number details on the Monitor tab of MaxAgent and
MaxCommunicator, check the Block Caller Name and Number box.
To add members to the list,
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Setting Up Monitor Lists
1.
From the Monitor Available list, select the extensions to add to the extension user’s MaxCommunicator
Change Monitor window.
2.
Click Add to move the extensions to the Monitor List.
To remove members,
1.
Select the extensions in the Monitor List.
2.
Click Remove.
Check the Trunk Monitor Enable check box to allow monitoring of the AltiLink Plus trunk events at the selected extension.
Click the Default button to return the settings to the default – the extension can monitor its own calls.
About Viewing Remote Extension Activity
In order for agents in MaxAgent and MaxCommunicator to see the activity for remote extensions on the
Directory tab, those agents must add the remote extensions to their Monitoring lists.
If they do not add the remote extensions to their Monitoring list, they will still see the remote extensions, but they will not see any activity for them.
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H A P T E R
15
Setting Up IP Extensions
IP phones communicate with the system using SIP protocol to establish the signaling channel and media channel (the voice steam, using RTP protocol). With SIP implementation, the system establishes a signaling channel to an IP phone when the IP phone is in use.
Figure 129. Concept of signaling and media channels
The media channel (voice stream) is connected between two IP phones under normal operation. There are some special situations that require you to configure the IP phone to connect its voice stream to the server.
Please see “Setting an IP Extension” on page 213 for information.
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Figure 130. Signaling and media channel between two IP phones
Signaling Channels
A SIP signaling channel communicates between the system and the IP phone to perform call control, including call setup, tear down, registration, and phone feature access.
The signaling channel implementation consists of the following elements:
•
SIP Virtual Board – Establishes a logical board ID relationship with other types of physical boards in the system (displayed on Board View window as SIPSP board).
•
SIP Signaling Channel – Creates SIP signaling channels for IP Extensions (access through SIPSP board, Channel Group configuration).
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Signaling Channels
•
SIP Extension Channel – Establishes a logical channel relationship with other analog and MobileExt ports (displayed on the SIPSP board configuration, Channel Mapping List).
•
SIP Extension Channel Activation – Associates an extension with a SIP Extension channel when IP phones register to the system (displayed in the Extension View window).
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Media Channels
A media channel is an RTP channel that connects system-to-phone, or phone-to-phone, system-to-system to carry the digitized voice stream. The codec resource on the VoIP board will be allocated dynamically based on connection types. If both end devices are IP phones, the media channel can be connected from IP phone to
IP phone using the IP phone’s codec, except when the following is true:
•
SIP trunk is used
•
Codecs at two end devices are mismatched
•
Extension has Agent setting checked
•
Voice recording is enabled at the IP extension
•
A NAT router exists between MAXCS and remote IP phone
•
SIP supports a direct connection of the voice stream between IP phones.
The media channel implementation consists of the following elements:
•
Configure Codec Profile – Creating a profile for each codec type, jitter buffer, packet length, DTMF tone delivery, and ring back tone treatment (SIP Early Media).
•
Assign Codec to Device – Configuring codec profile to a single IP address or a range of IP addresses.
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Setting an IP Extension
•
Monitor Codec Usage – Viewing codec usage status.
Setting an IP Extension
To make an extension an IP extension:
1.
In the Extension Configuration General tab, select the extension from the list at the left and check the
Enable IP Extension check box.
2.
Select the address type.
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Figure 131. IP Extension options
•
Dynamic IP Address – The system will associate the IP address to the extension when the IP phone registers automatically, or when the user logs on using #27+Enter from the AltiGen IP phone. This is the recommended setting.
•
Static IP Address – You need to enter the IP address for each IP extension. This setting is recommended only when connecting to third-party SIP devices such as a Multi-Tech MVP VoIP gateway with FXS ports support. (Refer to the MultiTech Gateway Application Note on the AltiGen web site.)
3.
Configure the rest of the IP Extension panel:
•
Connect Voice Stream to Server – The IP phone will always connect the media channel to the server when this box is checked. This box is checked by the system in the following situations:
–
The non-workgroup call recording option is checked for this extension.
–
This IP extension is a workgroup agent and the workgroup recording is checked.
–
You allow a workgroup supervisor to barge-in, listen to, coach, or record this agent's conversation.
•
Home Media Server ID – This configuration is meaningful for a gateway Softswitch system. When multiple chassis are configured to be a single system, you need to assign IP extensions to the configuration’s Home Media Server to be able to use its resources for activities such as the following:
–
Access voice mail
–
Initiate a conference call
–
Record a conversation
–
Barge in, listen, and coach by workgroup supervisor
Guidelines:
–
If the Softswitch and HMCP Media Server are in the same server, the default ID “00” will be the Home Media Server ID. No change is required.
–
If the HMCP Media Server and Softswitch server are separated, you need to assign IP extensions to the HMCP Media Server ID.
–
If you have two or more HMCP Media Servers, you need to assign each IP extension to one of them, based on resource usage.
•
Enable Polycom or 3rd Party SIP Device – If the extension is a Polycom phone or another SIP device, check this box. You must have a license for each 3rd party SIP device.
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Setting an IP Extension
•
3rd Party SIP Registration Password – Polycom phones and other standard SIP 3rd-party IP phones require a SIP Registration password to register the phone to the MAXCS server. The password should be the same as the password that is configured in the Polycom IP phone configuration file. This password is used only for phone registration purposes.
Note:
For Polycom phones, several password fields in earlier releases of MAXCS have been consolidated; refer to the MAXCS Polycom Configuration Guide for details.
•
Enable Fallback to Mobile Extension – When this option is checked, and the IP phone loses its network connection, it will automatically fall back to a Mobile Extension. The mobile extension channel must be specified from the list. This feature is only available for an IP Extension with a dynamic IP address.
Note:
In Release 8.0, Polycom phones also support the fallback feature.
Losing network connection can happen in the following cases:
–
The user presses #26 to log out from the AltiGen IP phone
–
The server loses connectivity to the IP phone
–
The IP Extension's channel is taken over by another extension
–
The user exits from an IP Talk session
Users may need to re-register when a phone falls back to a Mobile extension.
Once associated with a fallback mobile extension, when the network connectivity is restored, the fallback mobile extension stays active, and the user must re-register the phone (#27) to reconnect to the server. For Polycom phones, #17 can be used to re-register the phone.
•
Enable Fax-Over-IP – Enables the FoIP capability for this extension. Refer to Fax-over-IP
.
Setting VoIP Codec for IP Extension
The system has a pre-configured IP range and codec settings to assist IP phone deployment.
In Enterprise Manager, click the Codec button. In the Codec list, three codec profiles are pre-configured:
•
G.711 Mu-Law
•
G.729
•
G.723.1
In Enterprise Manager, click Servers > IP Codecs tab. Three local IP address ranges are pre-configured to use the G.711 codec profile:
•
192.168.0.0 ~ 192.168.255.255
•
10.0.0.0 ~ 10.255.255.255
•
172.16.0.0 ~ 172.31.255.255
When an IP phone registers to an IP extension, the system will check the IP address to determine which codec to use for the IP phone.
Also see HMCP Codec Preference on page 130 for a discussion of the Codec Preference feature.
For Local IP Phone Deployment
If your local IP address is not in the pre-configured range, you need to add the local IP address range into the
IP Codec setting. Otherwise the system will use the Default (Prefer G.723.1 support G.729) setting for your IP extensions.
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For Remote IP Phone Deployment
If you do not enter the remote IP phone's IP address into the IP Codec table, the system will use the Default
(Prefer G.723.1 support G.729) setting. You can change the Default to Prefer G.729 support G.723.1, if desired.
Assigning Codec Profiles to IP Addresses
Setting Fax-Over-IP for an IP Extension
MAXCSsupports T.38 Pass-through on systems using AltiGen SIP trunks.
T.38 is a standard protocol for real-time fax transmission over IP networks. T.38 Pass-through relays two T.38 sessions between the SIP trunk and MAXCS and MAXCS gateway.
•
T.38 pass-through is supported only on Softswitch and MAXCS Cloud
•
T.38 pass-through is supported only on systems using AltiGen SIP trunks
For FoIP configuration instructions, see Fax-over-IP Configuration.
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IP Phone Configuration
This chapter discusses both AltiGen IP phones and Polycom IP phones. For instructions on configuring
Polycom IP phones, refer to the MAXCS Polycom Configuration Guide.
The system administrator can control and program the following areas for each type of AltiGen IP phone (some settings apply to Polycom phones as well; refer to each specific section for details):
•
Specify the server IP address that the IP phone needs to register
•
Protect the IP phone configuration with a password
•
Prevent the user from changing the configuration from the IP phone
•
Configure the Trunk Access (Route Access) code
•
Configure the time zone and time format
•
Specify the TFTP server for firmware updates
•
Force the IP phone to reset and download new firmware
•
Set SIP transport settings for SIP security
•
Enable SIP telephony service for a selected third-party SIP device
•
Configure programmable keys
•
Allow the IP phone to receive workgroup real time status
•
Allow the phone to auto-discover the server’s IP address
Environments with both Polycom and AltiGen IP Phones
If your environment includes both Polycom IP phones and AltiGen IP phones, be aware that while Polycom phones support the G.722 codec, AltiGen IP phone models IP-705, IP-710, and IP-720 models do not support that codec. (AltiGen's IP-805 model does support G.722.)
Remote sites running Polycom and AltiGen IP phones can coexist even with only one public IP address, as long as you configure one codec profile with the following settings:
•
DTMF must be set to RFC 2833.
•
You must configure a list of codecs that include at least one of the following: G.711, G.723 or G.729. You cannot configure G.722 as the only codec. You can include G.722 in the list, as long as you have at least one of those other codecs in the list as well.
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Configuring the AltiGen IP Phone
Note: You will need to upgrade the firmware on the AltiGen IP phones to the latest release, so that the phones will work with MaxCS.
To configure the AltiGen IP phone, select PBX > AltiGen IP Phone Configuration.
This opens the AltiGen IP Phone Configuration window, where, after setting up an IP extension, you can set parameters for the extension.
This panel applies only to AltiGen IP phones
This panel applies to both
AltiGen and Polycom IP phones
Figure 132. IP Phone Configuration General tab
The left side of the IP Phone Configuration window displays all the IP phone extensions that have been set up in the system. The status "Inactive" means the Enable IP Extension box is checked for this extension in the
Extension Configuration window, but there is no IP phone logged in to the extension. The extension may be a physical extension using an analog phone, a MobileExt, or a virtual extension.
After creating the IP extensions, you can set the parameters on the General tab.
Note: The Apply To button works with the following parameters: General, TFTP Server field (Reset IP
Phone and Boot Download options cannot be applied to multiple extensions), Debug, and Display
Workgroup Status (IP 705, IP 805, and IP 600 phones).
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AltiGen IP Phone Parameters
The fields in the left part of the window apply only to AltiGen IP phones.
AltiGen IP Phone
Parameter
Description
General Info
Default Trunk Access
Code
Debug
TFTP
Network Setting
Lets you specify the IP address of the MAXCS system the IP phone is connected to. Also
see Configuring Auto-Discovery of Server IP Address on page 223. The version of
firmware associated with the IP phone is automatically displayed in the Version field.
To protect the configuration on the IP phone, check the Enable Protection on Menu check box and assign a numerical password. When the user presses the Menu button on the IP phone to access the phone configuration menu, the user will need to enter the assigned password. You can use this check box for two purposes:
•
If you publish the configuration password to the user, only the phone user would be able to change the phone configuration.
•
If you do not publish the configuration password, you can block the phone user from changing the phone configuration.
To protect only the E911 Location ID assigned to this phone, check the Enable Protection
on E911 checkbox and enter the password that the user must type on the device in order
the change the E911 configuration. See the chapter Location-Based E911 (for Roaming
on page 307 for instructions on configuration E911 for IP phones.
Lets you set the digit required to enable a user to return an outside call from the Call Log.
The default trunk access code can be the route access code, if it is set in
MaxAdministrator.
This is for debugging the IP phone using Telnet.
You must enter a Diagnostic password when logging in to MaxAdministrator (before you enter your Admin password) to enable this configuration.
Lets you assign the TFTP server to which the IP phone can connect for updating firmware when necessary. Enter the IP address of the TFTP server in the Server field.
To reset the phone and download the latest firmware image, check the Reset IP Phone and Boot Download check boxes. If you only check the Boot Download box, the firmware will be downloaded when the IP phone reboots (power cycles) next time.
Note: Make sure the TFTP server is running and the new firmware image is loaded to the correct directory before you reset and download firmware.
•
TOS (Hex) – Type of Service. 8 bits in the IP header are reserved for the service type.
They can be divided into 5 subfields: The 3 precedence bits have a value from 0 to 7 and are used to indicate the importance of a datagram. Default is 0 (higher is better).
Bits 3 4 5 represent the following:
D: requests low delay
T: requests high throughput
R: requests high reliability
•
Enable VLAN – If your network administrator has configured VLAN, check this check box to enable VLAN for the selected phone. Then enter the VLAN ID for the line port
(voice service) and the VLAN ID for the PC port (data service). (Get these IDs from
your network administrator.) See “Virtual LANs” on page 315 for information on VLANs.
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General IP Phone Parameters
The fields in the right part of the window apply to both AltiGen IP phones and to Polycom IP phones.
General IP
Phone
Parameter
Info
Time Display
3rd Party SIP
Device
Description
•
Shows the current version of firmware that is loaded on the phone. This field is readonly.
•
Offset
–
A per phone-based configuration that allows a remote AltiGen or Polycom IP phone to display a different time, based on location. The offset is the time difference, in hours, between the AltiGen system and the IP phone.
•
Format
–
A per-phone-based configuration that allows the AltiGen or Polycom IP phone to display the time in one of the following formats: 24 hour (example: 13:15), 12 hour AM/PM (example: 1:15 PM), or (on AltiGen IP phones) AM/PM 12 hour (example:
PM 1:15
)
•
Enable SIP Telephony Service – Enables SIP hold, SIP transfer, and SIP server-side conference features for the selected 3rd party IP phone extension.
If the IP phone is SIP-enabled, the Flash key (which includes the Hold button in
MaxAgent/MaxCommunicator) is not supported when you check this setting.
•
Enable Polycom Advanced Features – Enables additional features for various models of Polycom phones. Refer to the MaxCS 8.0 Polycom Configuration Guide for full details.
This parameter must be enabled for Polycom phones.
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General IP
Phone
Parameter
Description
SIP Transport
NAT Setting
These settings secure the SIP signaling messages and the RTP. SIP signaling is secured using transport layer security (TLS). RTP or SIP-associated media is secured using the secure RTP (SRTP) protocol.
•
Persistent TLS – Check this setting to have the selected extension communicate using TLS. The TLS protocol allows applications to communicate across a network in a way designed to prevent eavesdropping, tampering, and message forgery. TLS provides endpoint authentication and communications privacy for VoIP systems using cryptography.
If either side initiates SIP messaging with an alternate transport like UDP or TCP, these are supported, as well.
•
SRTP – Check this setting to have the selected extension use SRTP. SRTP is a version of RTP that provides confidentiality and message authentication. Since the SRTP session key is sent in the SIP signaling via SDP, the key can be exposed to eavesdropping. So SRTP needs to co-exist with TLS for the communication to be fully secure.
If SRTP is checked, the voice stream always goes through the server.
Changing the TLS/SRTP parameter settings for a Polycom phone will require rebooting the phone; otherwise the phone may not register with MAXCS.
If the IP phone is behind NAT, UDP will be used even if TLS and SRTP are checked, since
TLS cannot penetrate NAT.
IP Phone Configuration vs Enterprise Manager configuration:
SIP calls from one AltiGen server to another go through a SIP Tie Trunk. Configuring TLS
for this scenario is done in Enterprise Manager. See the section Setting VoIP Codec
.
Extension level policy has priority over the codec profile policy.
If the IP extension supports TLS and the codec profile set in Enterprise Manager does not, then the IP extension policy holds. That way you can configure a range of IP addresses in the IP Dialing table or IP Codec screen, and have only a few IP addresses/extensions support TLS.
If the IP extension does not have TLS configured as its transport, but the codec profile supports TLS for that extension, then the codec profile policy holds.
This setting is for a remote IP phone with a private address and behind NAT. When connecting to the AltiGen system, the system will use this information to execute the NAT traversal for the IP phone. The NAT status and address are read-only fields.
•
NAT Status
–
Indicates if the IP phone is behind a NAT router. Read only.
•
NAT Address – This is the NAT router’s public IP address, as set in the Extension
Configuration window. Read only.
Registry Keep-Alive Duration
–
Indicates how often a SIP registration message is sent to the server when the IP phone is behind a NAT router. You need to enter a Diagnostic password when logging in to MaxAdministrator (before you enter your Admin password) to enable this configuration. Default setting is 60 seconds.
Configuring Programmable Keys and Workgroup Status
After setting parameters on the General tab, go to the tab that corresponds to the phone type, and configure the programmable keys (plus the Display Workgroup Status field on the Alti-IP 600, IP 705, and IP 805 phones). Programmable key settings are described in the next table.
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Figure 133. IP Phone Configuration window, IP 805 tab
Note: The Copy From button allows you to copy Programmable Key settings from one IP phone extension to another. No other settings are carried over.
Parameter
Programmable
Keys
Description
Use the list to assign one of the following functions to the desired keys:
•
N/A
–
When selected, the corresponding programmable key cannot be used.
•
BLF (Busy Lamp Field)
–
When selected, enter an extension number in the field below; this will be associated with the corresponding programmable key to this extension number; the light in this programmable key indicates that the extension number is busy or ringing. You can select the Play Beep Tone check box to also have the IP phone play an audible beep or one of several different ring tones when the extension number is ringing.
Note: The BLF feature can be assigned only to internal extension numbers, not outside numbers.
See the MaxCS 8.0 Polycom Configuration Guide for details on configuring BLF keys for Polycom phones.
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Parameter
Programmable
Keys
Description
•
Feature Code
–
When selected, enter a MAXCS feature code in the field below; this will be associated with the corresponding programmable key to dial this feature code.
•
Admin Defined # – When selected, this programmable key can be configured by the administrator only. Enter a valid number 0~9, *, #, or F (Flash) in the field below.
One use for this can be to tag a call with an account code by pressing one button. For example, entering F#321 in programmable key 1 will cause a connected call to be tagged with account code 1 (F is for Flash, #32 is the extension feature code, and in this example, 1 is the account code). Account codes are set up in System Configuration, Account Code tab.
•
Line Park – When selected, enter the Line Park line ID in the field below. The user can press this programmable key to park a call or to retrieve a parked call.
See the MAXCS Polycom Configuration Guide for details on configuring Line Park softkeys for Polycom phones.
•
Call Record – When selected, the user can press this programmable key to start conversation call recording. This only works for extensions with Record on Demand selected in the Extension Configuration window.
•
WG Status – (IP710 only) When selected, the user can press this programmable key to see the real-time workgroup status (callers in queue, longest queue time, number of callers who have waited longer than the service threshold, and service level).
•
User Defined # – (Default) Allows the user to define the programmable key from the
IP phone.
•
Headset
– (Alti-IP 600, IP 705) When configured from the list for programmable key
10 (Alti-IP 600) or programmable key 5 (IP 705), the IP phone user will be able to activate a third-party headset (certified by AltiGen).
•
Flash
– (Alti-IP 600) Upon initial installation, the lower left programmable key is set up as FLASH by default. This key can be re-assigned in MaxAdministrator, using the
AltiGen IP Phone Configuration window. No other programmable keys can be configured to FLASH.
(Alti-IP 600, IP 705, IP 805) When enabled, allows the IP phone to display workgroup queue status, such as number of queued calls, the current longest queue time, agent login/logout state by pressing the Down arrow key.
Display
Workgroup
Status
Important:
The configuration in MaxAdministrator will override the IP phone’s local configuration after the IP phone is registered. If the IP phone’s local configuration is changed while in Basic mode, these changes will be overwritten by MaxAdministrator settings.
Administrators should perform any updates to the IP phone’s firmware after normal business hours or when the IP phone is not in use. If the IP phone is in use during an update, not only will the call will be disconnected, but if the IP phone is powered off by the user during the firmware upgrade, the IP phone
may become unusable
.
Configuring Auto-Discovery of Server IP Address
You can configure option 120, in your DHCP server with your MAXCS IP address, so that the AltiGen IP phone automatically discovers the MAXCS server IP address and only needs to have the extension and password entered.
In addition to making initial IP phone setup easier, this feature is also helpful when there is a need to migrate
MAXCS to a new IP address. The administrator just needs to update the new MAXCS IP address in the DHCP server and then reboot all AltiGen IP phones. The phones will automatically pick up the new MAXCS IP address.
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Warning! In the event that there are two MAXCS servers in a same network and all IP phones get their IP address from a single DHCP server, some IP phones will get the wrong server IP address. You need to disable the auto-discovery feature for those IP phones that log on to the MAXCS server that is not configured in the DHCP option 120.
Setting Up DHCP Option 120
Different DHCP servers have different ways to set up options. The following example uses Microsoft Windows
DHCP Server to define option 120. Since option 120 is not available by default, you must create it.
1.
Open the DHCP configuration window.
Right-click the server and select Set
Predefined Options
2.
Right-click the server and select Set Predefined Options.
Click Add
3.
Click the Add button.
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Configuring Auto-Discovery of Server IP Address
4.
Enter the following:
Name: AltiGen Server IP Address
Data Type: String
Code: 120
Description: AltiGen Server IP Address
5.
Click OK twice.
6.
Under the DHCP scope you created is a field labeled Scope Options. Right-click Scope Options and select Configure Options.
7.
Check option 120.
8.
Enter the IP address of your MAXCS server in the String value field.
9.
Click Apply and OK. The scope now shows option 120.
Check option 120 and enter the IP address of your MAXCS server in the String value field
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10. Right-click the scope option 120 and select Activate to activate the scope.
On the AltiGen IP Phone
The IP phone's System menu includes an item called Auto Discovery. The user can select YES or NO for this menu item. The factory default is YES.
When you Upgrade Firmware
•
When you upgrade from firmware that does not support Auto Discovery, Auto Discovery will be disabled by default.
•
When you upgrade from firmware that does support Auto Discovery, the Auto Discovery setting will carry over.
•
When the user erases the IP phone configuration by using **2 [enter] in the IP phone menu, Auto
Discovery will be enabled by default.
Possible scenarios
•
During the AltiGen IP phone’s start-up stage, if Enable DHCP is ON and Auto Discovery is set to YES, the IP phone configures its IP address from DHCP, and at the same time, it gets the MAXCS SERVER address from DHCP option 120. The user is then prompted to set his extension number and password.
•
If Enable DHCP is OFF, then the phone’s IP address and the MaxCS 8.0 SERVER address must be set manually.
•
If Enable DHCP is ON and Auto Discovery is NO, the DHCP option 120 value is not sent to the AltiGen
IP phone. The MAXCS SERVER address must be set manually.
•
If Enable DHCP is ON and Auto Discovery is YES and DHCP option 120 is set, the AltiGen IP phone always gets a new IP address, and DHCP option 120 refreshes the value of MaxCS 8.0 SERVER, even if MAXCS SERVER already has a value. The screen pauses for 2 seconds while the IP phone gets the
MAXCS IP address from DHCP 120.
Disabling Auto-Discovery
To disable auto-discovery on individual AltiGen IP phones, each phone must have its Menu > System > Auto
Discovery menu item set to NO.
To disable auto-discovery on all phones, do not set DHCP option 120, or delete it if you have already set it.
When auto-discovery is disabled, the MAXCS SERVER address must be set manually.
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Polycom Configuration
When You Have Two AltiGen Servers in the Same Network
If there are two AltiGen servers in the same network, some AltiGen IP phones will get the wrong server IP
Polycom Configuration
For instructions on using the Polycom tab to configure Polycom phones, refer to the MAXCS Polycom
Configuration Guide
.
Figure 134. The Polycom tab of the IP Phone Configuration page
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17
Mobile Extension Configuration
If your company has employees working at home or servicing customers in the field, you can connect their home phones or cell phones to the AltiGen PBX, providing them with the same productivity features as if they were working in the office.
AltiGen's ExtensionAnywhere capability allows an extension/agent to be:
•
On-premise using voice or data wiring
•
Mobile or remote using IP phone, cell phone, or PSTN phone
•
An extension of another PBX via adjunct tie trunk or over a PSTN trunk simulated as a mobile extension port.
When configured, the property of the trunk interface is changed to simulate an extension. A mobile extension user will gain most of the system routing, call control, voice mail, CTI, and call center features through the
PSTN telephone network.
A mobile extension includes the following capabilities:
•
Call control – Transfer, hold, park, call pickup, conference
•
Call handling – Single/multiple call waiting and queuing, RNA routing, account codes
•
MaxCommunicator, MaxMobile Communicator, and MaxAgent CTI clients
•
Conversation recording
•
Workgroup agent with login/logout and ready/not-ready
•
Pressing ** terminates a call (soft on-hook) and gets a dial tone for the next call. The second * must be pressed within 1.5 seconds, or the system interprets it as one *.
•
#82 – Dial tone mute
•
Supervisor silent monitoring, coaching, and barge-in
The extension can be dynamically logged in using #27 from an internal, mobile, or IP device.
MobileExtSP Board Overview
A simulated physical board (MobileExtSP board) is created when you install the MAXCS Softswitch. You can configure this board with up to 1000 mobile extension ports. It handles all system-wide mobile extensions.
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Chapter 17: Mobile Extension Configuration
Figure 135. MobileExtSP board diagram
T1, PRI, analog, and SIP trunks can be shared for regular incoming and outgoing calls and mobile trunk connections.
A mobile trunk can be assigned a Group ID and mobile extensions can be assigned to the appropriate group.
An analog trunk can be dedicated to one mobile extension user. A PRI trunk and SIP trunk can only be shared by all mobile extension users.
Configuring the MobileExtSP Board
To configure the MobileExtSP board,
1.
In the Boards window, double-click the MobileExtSP board. In the Board Configuration window, doubleclick a channel group.
Figure 136. Opening the Mobile Extension Board Configuration dialog box
The Mobile Extension Board Configuration dialog box opens.
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Configuring the MobileExtSP Board
2.
Click the Add/Remove button to add mobile trunks.
3.
Add trunks to the Mobile-trunk Member List from the Not Member List by selecting the channels and clicking the Left Arrow button. You can use the Shift key or Ctrl key to select multiple channels.
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You need to assign a Group ID to the channels. This Mobile Trunk Group ID allows you to differentiate
MobileExt users connecting through different trunk types, like PSTN, SIP, or cell phone gateway. You can assign a mobile extension to use a specific trunk group. For example, if you assign SIP trunk channels from 1-3 to Group 001, and mobile extension 237 is assigned to Group 001, then when you make a call to extension 237, only the SIP Trunk channels from 1-3 can be seized. If all three channels are busy, the call will fail while other mobile extensions using another mobile trunk group ID may not be impacted.
Mobile extensions are assigned to a group in the ExtensionAnywhere Configuration dialog box (see
Note: If a PRI span is used, only the whole span can be added or removed, not individual PRI channels.
T1 and analog trunks are added or removed individually.
Although a whole PRI span is added, if Mobile Trunk Allocation is selected as Shared
(see Figure 138 on page 234), individual trunks, when idle, still can be used dynamically
by normal PRI trunk traffic or mobile extensions.
4.
On the left side of the Mobile Extension Board Configuration dialog box, configure the fields:
•
Max Number of Extensions – If more mobile channel support is required, change this to a larger number (1000 extensions maximum), and then reboot the system.
•
Key Simulation – Check the first check box to allow the mobile phone user to use the * key to simulate
“FLASH”. Check the second check box to allow the user to use ** to disconnect the current call and then get a dial tone without hanging up the cell phone. The user must press the second * within 1.5 seconds.
•
“Transmit Caller ID to MobileExt through PRI” panel. Choose from:
•
Send inbound caller ID or extension number
•
Send inbound caller ID only
•
Replace with following number
•
DNIS Access Numbers – If a PRI trunk is used for a mobile extension, a DNIS access number must be set, so that MAXCS can tell if the incoming call is a regular trunk call or a mobile extension off-hook request. Click the Add button in this panel to add a DNIS access number. To remove a number, select it and click the Remove button.
•
Mobile Extension Ports table – Displays fields for the channel, target phone number, caller ID, trunk allocation (shared or dedicated), phrase 1 (Play Phrase After Answered), and phrase 2 (Play Phrase
Before Dial Tone) of each extension port.
•
Mobile Trunks table – Displays fields for the board, span, channel, trunk allocation, mobile extension and status of each mobile trunk.
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5.
Note the logical ID of the MobileExtSP board. You will need it when you assign an extension to a mobile port.
6.
When you are finished adding channels as mobile trunks, restart MAXCS.
Configuring an Extension as a Mobile Extension
To configure an extension as a mobile extension,
1.
Open the Extension Configuration window.
2.
To assign an extension to a mobile extension port, select a virtual extension and change it to a physical extension.
Figure 137. Changing a virtual extension to a physical extension and setting the location
3.
By clicking the Next or Prev button in the Location panel, select the Logical Board ID of the MobileExtSP board and Logical Channel ID for this extension, then click Apply.
The new location is displayed in the Agent/Supervisor/Extension list.
4.
Click the Line Properties button to configure the mobile PSTN number and other options for the mobile extension. The ExtensionAnywhere Configuration – MobileExtSP dialog box opens. (Alternatively, from the Mobile Extension Board Configuration dialog box you can double-click the mobile extension port to open the ExtensionAnywhere Configuration.)
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For a mobile phone using
MaxMobile Communicator, clear all the Phrase check boxes.
Figure 138. ExtensionAnywhere Configuration – MobileExtSP dialog box
•
Name Enter the name of the person using the mobile phone.
•
Target Phone Number – Enter the number of the mobile phone. This is used when MAXCS makes a call through PSTN to the mobile phone. Do not include the trunk access code.
•
Caller ID – Enter the phone number of the mobile phone. This is for incoming caller ID verification.
MAXCS uses it to determine whether a call is from a mobile extension. If the caller ID is matched, the mobile extension user will hear a dial tone from the system, the same as an internal extension user hears when the phone is off-hooked.
It's also used to find a mobile channel in the MaxMobile Communicator application, and it is used in the
MaxMobile Communicator login.
Note: When a MaxMobile user logs in to the MAXCS system, the assigned extension number, extension password and cell phone number are used as identification. First, MAXCS checks the extension number and extension, then it uses the cell phone number to search the mobile channel table. If
MAXCS finds one channel's Caller ID is the same as the cell phone number, it will assign this channel to the extension number. The extension is allowed to log in as a mobile extension. If no channel is found, the login fails.
•
Mobile Trunk Allocation – Select either Shared or Dedicated.
Shared – When selected, this mobile extension will share mobile trunk ports with other mobile extension users. You need to assign a mobile trunk Group ID to this extension. The system will dynamically allocate a mobile trunk port within this Group ID when the system calls out to this mobile extension number.
When the mobile extension user calls into the system, any mobile trunk port can answer the call, verify caller ID, and play a dial tone to the mobile extension user.
Dedicated – Only analog trunks can be dedicated mobile trunks. When selected, you need to assign a mobile trunk port to this mobile extension. You have the option to disable caller ID verification if a mobile trunk port is dedicated to this mobile extension. The mobile extension user will hear a
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Configuring an Extension as a Mobile Extension dial tone when calling into this specific trunk port. Use the Browse button (…) to select the desired mobile trunk.
Figure 139. Mobile Trunks dialog box
•
In the Phrase panel, you have three options: You can select either Press Any Key To Answer Call
or
Play Phrase After Answered. And you can select Play Phrase Before Dial Tone. You can use the
Apply button to apply selections in this panel to other mobile extensions.
Note: Note: For mobile extensions running MaxMobile Communicator, you should clear all three check boxes in the Phrase panel.
–
Press Any Key To Answer Call – When a call is answered by this mobile extension user, the system will play the following phrase for the mobile extension user: “To accept this call, please press any digit.” The user must press any key within 3 seconds to connect the call; otherwise, it will time out and the call will be treated as an agent/extension RNA and will be routed according to its workgroup/extension setting.
–
If there is a network error or a mobile extension trunk is not available, RNA handling is applied to the caller. Therefore, it is suggested that you don’t check the Set RNA Agent Logout option for the group that contains the mobile extension as an agent (Workgroup Configuration, Call
Handling tab).
–
Play Phrase After Answered – The system will play the given phrase when the mobile extension user answers the call from the system. The default phrase (9037) is a special tone to signal the mobile extension user that this call can be put on hold, parked, transferred, conferenced.
–
Play Phrase Before Dial Tone – The system will play the default phrase 9037 (a special tone) and then the dial tone when the mobile extension user calls into the system through a configured DNIS Access Number.
Additional Configuration for MaxMobile Communicator
For mobile phones running MaxMobile, do the following:
•
If MAXCS is behind NAT, configure the NAT router to forward TCP port 10080 and 10081 to MAXCS's private IP address, so the data access from a 3G network can reach this server.
•
Open firewall ports TCP 10080 and 10081 for both virtual public IP address and private IP address.
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•
Assign an AltiGen MaxMobile license to the extension. To do this, from the MaxAdministrator main menu, select License > Client SEAT License Management. In the Client SEAT License Management dialog box, select MaxMobile in the License Types column, and add the appropriate extension to the Members list.
•
When using a SIP trunk as MaxMobile trunk, the Early Media option must be enabled for the SIP trunk.
Voice Mail for Mobile Extensions
When the mobile extension phone is turned off or busy, messages can go to the extension’s voice mail in MAX-
CS or to the mobile phone’s voice mail:
•
To send a call to the mobile extension’s voice mail in MAXCS, check the Press any key to answer call check box. If the mobile phone is running MaxMobile Communicator, this check box should never be checked.
•
To send a call to the mobile phone’s voice mail, the Press any key to answer call check box must be
un
checked.
Mobile Extension Limitations
•
Only PRI mobile trunks can deliver Caller ID information to the mobile extension.
•
A mobile extension cannot support Centrex transfer.
•
After adjusting the number of mobile extension ports in a mobile extension board, MAXCS must be restarted for the changes to take effect.
•
Cannot deliver caller name to the mobile extension.
•
Does not support Message Waiting Indicator on the mobile extension device. (Use Message Notification as a work-around).
•
Since the DTMF key * is used for simulating the FLASH signal, there is no way to send * to the system.
•
The RNA for mobile extension may not be accurate, because the system ring count may not be in sync with the mobile extension device ring count.
•
When placing calls to mobile extensions that are cell phones, if the cell phone is out of signal range, the caller may hear long periods of silence. You can check the Press any key to answer call option to prevent this problem.
•
Only analog trunks can be allocated as dedicated mobile trunks.
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Hunt Group Configuration
The hunt group is a simple call distribution application for operator, call coverage group, integration with a fax server, or a user with multiple extensions connecting to different devices. When adding a member to a hunt group, the following rules apply:
•
No agent seat license required
•
Any extension can be added to a hunt group
•
Each hunt group can have up to 128 members
•
An extension can belong to multiple hunt groups
Although a hunt group has call queuing capability, it lacks the following functions:
•
Does not generate real-time queue and agent status for the hunt group
•
Does not have a real-time counter to track hunt group activities for reporting purposes
•
Does not have logout reason code tracking capability
•
Does not have recording capability
•
Does not have service level threshold setting
•
Does not have queue overflow and quick queue option
•
Limited call distribution capability
•
No supervisor application to manage agents and calls in queue
•
No client application for agents to perform login/logout
The Huntgroup Configuration window provides for creating hunt groups, setting their attributes, and assigning group members.
To open the Huntgroup Configuration window, select PBX > Huntgroup Configuration.
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Figure 140. Huntgroup Configuration window
Overview of Huntgroup Configuration Window
These are the tabs in the Huntgroup Configuration window:
•
General – Add or delete a hunt group, assign a group name, password, and DID number
•
Group Member – Add or remove members from huntgroups
•
Mail Management – Capacity and feature options for hunt group mailboxes
•
Notification – Preferences and options for voice mail notification
•
Call Handling –Call forwarding, call waiting, and call handling preferences and options
•
Queue Management – Options for setting default or custom phrases used as queue announcements
Apply to Button
The Huntgroup Configuration window often allows you to apply changes to a particular hunt group or to select many huntgroups to which to apply the changes.
The Apply to button is disabled unless there is a change that can be applied to multiple hunt groups, and when you use it to apply changes to multiple hunt groups, it works on only those changed attributes that can be applied.
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Setting Up Huntgroups
Set up new huntgroups in the Huntgroup Configuration window.
To add a hunt group,
1.
Click the Add button under the Group List.
Setting Up Huntgroups
Figure 141. Adding a new Hunt Group
2.
Type in a group number for the hunt group.
3.
Check the Global group check box if you want the group to be visible to other systems within the VoIP
domain. See the section Enterprise VoIP Network Management for more information.
4.
Click OK.
Establishing Basic Hunt Group Attributes
After you create a hunt group, you can set basic attributes in the Huntgroup Configuration, General tab:
To set Hunt Group information, enter the following information:
•
First Name and Last Name – Each with a maximum of 32 characters.
•
Password – The default is the system default password set on the Number Plan tab of the System
Configuration window.
A valid password cannot be the same as its hunt group number and must be 4 - 8 digits (numbers or letters
A - Z) in length. Basic password patterns, such as repeated digits (1111), consecutive digit strings (1234), or digits that match the extension (Ext. 101 using 1012, 9101, 10101, etc.) are not recommended. The letters map to numbers (on a phone, for example) as follows:
Numbers
4
5
2
3
Letters
A, B, C, a, b, c
D, E, F, d, e, f
G, H, I, g, h, i
J, K, L, j, k, l
Numbers
8
9
6
7
Letters
M, N, O, m, n, o
P, Q, R, S, p, q, r, s
T, U, V, t, u, v
W, X, Y, Z, w, x, y, z
•
DID Number – Each hunt group can be assigned a DID number. This number does not have a fixed length, but the length must be long enough (range 2 - 16) for the system to match the DID incoming call.
•
Enable Dial-By-Name Service – Check this box to allow callers to search the list by employee name for this hunt group extension.
•
Description – Describe the purpose of this hunt group.
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Setting Call Restrictions
The call restriction rules on the General tab apply to users making outbound calls from within voice mail and several hunt group settings. These settings do not impact the call restriction settings configured for the hunt group member's extension in Extension Configuration.
•
Allow Calls to be Transferred or Conferenced to an Outside Number – When checked, the internal extension user can log into this hunt group voice mail, make a call to a second party, then transfer or conference to a third party.
•
Allow User to Configure Forwarding, Notification, and Reminder Call to an Outside Number – This setting regulates hunt group call forwarding, voice mail notification, and reminder call configuration. If this setting is not checked, you will see a warning message pop up when trying to set up forwarding to an outside number. International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Return Calls from within Group's VM System – When checked, an outside caller can dial into the system, log in to hunt group voice mail, and make or return calls from the group's voice mail (Zoomerang feature). International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Forward International Calls from within the Group's VM system –
This setting regulates making international calls from voice mail and forwarding to an international number.
Important:
Allowing any of these options may increase the potential for toll fraud. Make sure the password is properly configured to prevent an intruder from using this voice mail box to make an outbound call. AltiGen recommends that you leave the fourth option unchecked for all huntgroups at all times.
Establishing Hunt Group Membership
There are two ways to assign extensions to huntgroups.
•
In the Huntgroup Configuration window select a group, then click the Group Member tab. Here you can add extensions (group members) to the selected hunt group.
•
In the Extension Configuration window select an extension, then click the Group tab. Here you can assign a hunt group to the selected extension (and you can see what other huntgroups the extension is a member
of). For this second method, see “Adding or Removing Group Assignments” on page 189.
like to adjust and use the Up or Down button to change the order.
When you add an extension to a hunt group, the extension is in the "Logout" state. The hunt group member must manually log in using feature code #54.
Adding Extensions to a Hunt Group
1.
In the Huntgroup Configuration window, select the hunt group number in the Group List. The hunt group number appears in the window title bar.
2.
Click the Group Member tab.
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Figure 142. Huntgroup Configuration, Group Member tab
3.
Select the extension number(s) in the Not Member list. Use Ctrl+click or Shift+click to select several extensions.
4.
Click Add to move them to the Member list.
Note: If the hunt group pilot extension is configured to Ring All Available Members, the maximum number
of members is 20. See “Setting Call Handling Options for details.
Removing Extensions from a Hunt Group:
1.
Click the extension number(s) in the Member list.
2.
Click Remove to move them to the Not Member list.
Setting Login Status for System Restart
Whenever the system is restarted, the administrator can use the list at the bottom of the Group Member tab to:
•
Keep Login Status – All group members retain their original login status for that group prior to restart
(default setting).
•
All Login – All group members are automatically logged into the assigned group after the system is restarted.
•
All Logout – All group members are logged out of the group when the system is restarted.
Setting Hunt Group Mail Management
The Mail Management settings define how voice messages are handled for a hunt group, including how messages are announced and processed, and how much capacity is allotted to message storage.
To work with mail management settings, click the Mail Management tab, and select the hunt group number you want to work with from the Group List.
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Figure 143. Huntgroup Configuration, Mail Management tab
Note: You can use the Apply to option to apply mailbox settings to one, some, or all huntgroups.
Disabling a Mailbox
When you disable a mailbox, the normal greeting is played but callers cannot leave messages.
Setting E-mail Options
On the Mail Management tab, you can set the e-mail options for the hunt group:
•
E-mail Name – The hunt group’s e-mail name without the @domain. The default e-mail name is ext<hunt
group number>,
that is, the letters “ext” followed by the hunt group number. For example, the default email name for hunt group 500 would be ext500.
•
Retrieve Voice Mail by E-mail Client – When selected, this sends voice mail to the user’s e-mail as an attachment. Deselected, voice mail is retrieved as voice mail.
•
Enable Mail Forwarding – When selected, the hunt group’s e-mail will be forwarded to the e-mail address you specify in the Forward E-mail Address box. The address should be a full address, including the domain (for example, [email protected]).
If you enable mail forwarding, you also specify what you want done with the original messages after they have been forwarded. In the list you can choose to:
•
Delete Messages after Forward
•
Keep the Messages as New
•
Keep Messages as Saved
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Setting Mailbox Playback Options
You can use the following check boxes to turn on or off options for listening to playback of recorded messages.
These options apply to both new messages and saved messages, and they can be applied to multiple huntgroups using Apply to:
Message Playback
Parameter
Announce Message
Sender Before Playback
Announce Time Stamp
Before Playback
Confirm Callback Number
Enable Distinctive Call
Waiting Tone
Play the Newest Voice
Message First
Description
Selected, the user hears the name of the message sender
(internal sender only) before listening to recorded AltiGen
Voice Mail System messages.
Selected, the user hears the timestamp (time and date) of each message before playback.
Selected, system confirms the accuracy of the caller’s number.
Selected, the user hears three different call waiting tone cadences to distinguish between internal, external, and
operator calls (see “Distinctive Ring” on page 51).
Selected, new voice mail will be retrieved first. When not selected, the system will play voice mail based on FIFO (first in, first out).
Setting Mailbox Capacities
You can set various mailbox capacities with the following options, and you can apply the settings to multiple hunt groups using Apply to:
Mailbox Capacity
Parameter
Max Number of
Messages
Mailbox Size
Max Message Length
Retention Length of
Saved Messages
Description
Maximum number of messages stored in the hunt group’s mailbox. The range is 1 – 999, defaulting to 100.
Mailbox size in MBs of stored messages. The range is 1–500
MB, with a default of 50.
Maximum length of voice messages in minutes. The range is 1–
30 minutes, with a default of 5 minutes.
Number of days saved messages are archived by the system.
The range is 1–90 days, with a default of 60.
Setting Message Notification Options
To set notification options on new incoming e-mail and voice messages, click the Notification tab in the
Huntgroup Configuration window, and select the hunt group number from the Group List.
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Figure 144. Huntgroup Configuration, Notification tab
Individual users can also configure Message Notification within the AltiGen Voice Mail System.
Setting the Message Types for Notification
Select the types of messages for which the hunt group user will be notified:
•
None – When selected, the user is not notified with a call regarding newly received messages. Selecting this option does not prevent the user from getting message waiting indicators or stutter dial tone when new messages are received.
•
Urgent Voice Messages Only
•
All Voice Messages
Please note that the system will start notification as soon as it enters non-business hours under the following conditions:
•
Extension is set to notify Urgent Voice Message Only
•
Notification is set to Non-Business Hours
•
Voice mail is received during business hours and is marked urgent
•
Extension user does not check the urgent message
Setting the Type of Notification
There are four options for sending the notification or reminder message: phone, pager, extension or custom
application (Custom App).
•
Extension – To use the Extension option, select the Extension radio button, then type the extension number into the text box.
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•
Phone/Pager – For the Phone and Pager options, first specify the trunk or route access code using the list next to the Extension radio button. The Any option means to locate any available trunk. Then type in the number with all relevant dialing prefixes other than the trunk code, using a maximum of 63 digits.
•
Custom App – When used in conjunction with a third-party notification application, the Custom App feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log-on extension to which the third-party application is connected. Contact your local
AltiGen Partner for more information on using this feature.
Note: The Reminder Call will not work with this selection.
Note also the following considerations:
•
For the Pager option, the system calls the specified pager number and then dials the system main number
(as set in System Configuration, General tab), which is then displayed on the user’s pager.
For the operator-assisted paging function, the operator phone number and the pager number must be entered in the <phone number>*<pager number> format. For example, if the phone number to call the pager operator is 7654321 and the pager number to page the user is 12345678, the notification outcall number that needs to be entered is 7654321*12345678. When the pager operator answers the Message
Notification call, MAXCS announces the pager number and the System Main Number (as configured on the General tab of System Configuration), which will be displayed on the user’s pager. The operator is also given the option to repeat these numbers by pressing ‘#’.
Outcall to Cellular or PCS Phone Numbers
When an outcall is made by the system (for One Number Access, Message Notification, Zoomerang, Call Forwarding, and so on) to a cellular or PCS phone, it may ring the phone once but not necessarily present the call and make a connection. This will happen if the ringback tone played by the cellular service provider does not conform to standard ringback tones. To work around this problem, append a few commas (,) to the outcall (cellular) number when entering it. Each comma provides a one second pause.
Setting Notification Timing
When notification is configured to an outside phone number, the system will announce, "This is the outcall notification message for…" after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the carrier. If the system plays the announcement phrase before the notification call is answered, the phrase will be cut off. The following two options can be configured based on answer supervision capability:
•
Seconds after Dialing – If the carrier of the outside phone number cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
Note: If the delay is set too long, the notified party will hear silence before the announcement is played.
•
Seconds after Answered – This field is set to 0 seconds and it is not configurable for notification to a phone number. It means the system will play the announcement immediately after answer supervision is received.
When notification is configured to a pager, the system will transmit DTMF digits as the return phone number
(the System Main Number as set in the System Configuration General tab) after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the pager system. If the system sends digits before the call is connected, some digits will be cut off. The following two options can be configured based on answer supervision capability:
•
Seconds after Dialing – If the pager carrier cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
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•
Seconds after Answered – If the answer supervision signal is provided by the carrier, check this option and set the delay timer to 2 to 5 seconds. In some cases, the pager carrier cannot detect DTMF right after the call connection. (Default is 10 seconds, maximum is 30.)
Note: You may need to try a different delay setting to make sure the user return number is transmitted properly after configuration.
Setting Notification Business Hours
You can choose one of three options for when the extension user is to be notified of new messages:
•
Non-Business Hours – Notify only during non-business hours. Business hours are set in System
Configuration, Business Hours tab (see “Setting Business Hours” on page 56).
•
From/To – Notify during a specified time of day. Select the hours in the From and To time scroll boxes.
•
Any Time – Notify at all times (every day).
Setting Call Handling Options
Call Handling options include handling busy calls, forwarding, handling no-answers, call distribution, and other options.
Button” on page 238 for more information on using Apply to.
To work with hunt group call handling options, click the Call Handling tab in the Huntgroup Configuration window, and select the hunt group number from the Group List.
Figure 145. Huntgroup Configuration, Call Handling tab
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Handling Busy Calls
You have several options for handling calls while the agents in a hunt group are busy. If you do not enable busy call handling, the caller simply hears a busy signal.
To enable the options, check the Enable Busy Call Handling check box, then select from the following forwarding options:
•
Group Queue – The caller will stay in the hunt group queue waiting for any agent to become available. If there is no agent logged in at this moment, the system will use Group Logout Handling to handle this call.
•
Group Voice Mail – The caller will be forwarded to the hunt group voice mail box when all agents are busy
•
AA – Forward caller to an auto attendant.
•
Extension – Forward caller to an extension.
•
Group – Forward caller to another group.
•
Line Park – Forward caller to a Line Park group.
Forwarding All Calls
When you do not want the hunt group to handle any calls, check the Enable Forward To option in the Forward
All Calls section of the Call Handling tab, and select an option.
The forwarding options are as follows:
•
To Voice Mail
•
To an Extension – Select an extension number in the drop-down list.
•
To AA – Select the AA to use in the list under the option.
•
To a Group – Select a group from the list.
•
To the Operator
•
To an Outside Number – This option is available if it is allowed in the Other Call Restrictions option in
Cellular or PCS Phone Numbers” on page 245.
•
If you choose Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
•
To an App Ext – When used in conjunction with a third-party notification application, the App Ext feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log-on extension to which the third-party application is connected. Contact your local AltiGen
Partner for more information on using this feature.
•
To Line Park – If configured, select a Line Park group from the list.
Handling Unanswered Calls
The Enable No Answer Handling configuration provides options for handling calls when the system rings the first available agent and the call is not answered. If all agents in the hunt group are rung and no one answers the call, the system will use the Group RNA/Logout Handling rule. Enable No Answer Handling is not available if Intra Group Call Distribution is set to Ring All Available Members.
To configure this option, check the Enable No Answer Handling box.
Select one of the following forwarding options for no answer call handling:
•
Next Group Member – Ring the next available agent until all available agents are rung. If all agents are busy, caller will stay in the hunt group queue.
•
Extension – Take the call out of the hunt group and forward it to an extension.
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•
Group – Take the call out of hunt group and forward it to another group.
•
Group Voice Mail – Transfer the caller to the hunt group voice mail when the first available agent does not answer the call.
•
Member Voice Mail – Transfer the caller to the first available agent's voice mail if this agent does not answer the call.
•
AA – Take the call out of the hunt group and forward it to an auto attendant.
•
Line Park – Take the call out of the hunt group and forward it to a Line Park group.
If you select Ring All Available Members in the Intra Group Call Distribution section, then specify the Number
of Rings before Handling, using the scroll box beside that option. The number of rings is the total number of times agents are rung before the call is handled by the Group RNA/Logout Handling configuration
Setting a Hunt Group’s Call Distribution Rule
The Call Handling tab in the Huntgroup Configuration window lets you set the distribution of normal inbound calls to group members, using one of the following three options:
•
Ring First Available Member – First available extension in a hunt group. For example, if there are three member extensions in a hunt group, the call is always sent to the first member configured in the hunt group. If this member is busy, the call goes to the second member configured and so forth.
•
Ring Next Available Member – A round-robin method that attempts to evenly distribute calls among the group members. This method sends the call to the next member configured in a hunt group (regardless of whether the previous member is busy or not). In other words, if the previous call was sent to #3 in the group, the present call is sent to #4, if #4 is not busy.
•
Ring All Available Members – All extensions in a hunt group.
Note: When this option is enabled, a single hunt group can have no more than 20 members.
In addition, calls to the hunt group with this option enabled have higher priority than other hunt group calls. Therefore, if an agent belongs to multiple huntgroups, one of which has this option enabled, a call to that hunt group will be distributed before others, regardless of its Wait Time in the queue.
In addition, if you check the Enable Single Call Handling for Agent check box, the system will not send calls to an agent who puts a call on hold. If this option is not checked, the system will distribute calls to the agent even if the agent has a call on hold. In other words, this configuration determines if an agent can get multiple hunt group calls or not.
Handling Calls when Group Members Are RNA/Logged Out
You can set calls to forward to a specified destination when all group members either do not answer the call
(RNA) or are logged out. To do so, in the Group RNA/Logout Handling section of the Call Handling tab, check the Enable Forward to check box, and select a destination from the list. The forwarding options are the
same as for “Forwarding All Calls” on page 247.
Setting Queue Management Options
In the Queue Management tab of Huntgroup Configuration, you can specify which greetings and updates to use and you can set the update interval. For each hunt group you can either use the system default audio peripheral configuration or you can set up a custom configuration.
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About Fax-over-IP Hunt Groups
Figure 146. Huntgroup Configuration, Queue Management tab
tom configuration in the Queue Management tab involves selecting other available phrases from the lists. Depending on how long the caller is in the queue, the caller will hear phrases 1-5, in order, after which phrase 5
About Fax-over-IP Hunt Groups
MAXCS supports FoIP Hunt groups; there is no specific configuration type for such hunt groups.
In a FoIP hunt group, all group members must be FoIP extensions. Group queuing is not supported.
Limited FoIP group parameters are supported:
•
First/Last Name, Password, Description, Department
•
DID Number
•
Ring First Available Member and Ring Next Available Member
•
Busy Call Handling and No Answer Handling should be disabled
•
Forwarding All Calls – forwarding target must be a FoIP extension or FoIP Hunt group
Configure FoIP Hunt groups as you would any other type of Hunt group; include only FoIP extensions.
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Paging Group Configuration
The IP paging group is a group of IP phones that can receive station paging. This feature also can be used as
IP zone paging by creating multiple paging groups.
Note: Polycom paging groups are supported by the MAXCS Private Cloud service; AltiGen paging groups are not supported by the Private Cloud service.
Paging is limited to the local LAN.
AltiGen IP phone Implementation details:
•
Polycom paging groups are now supported. Polycom phones can page to AltiGen paging group but cannot receive pages from AltiGen paging groups. They can only receive pages from assigned Polycom paging
groups. See Polycom Paging Groups on page 253.
•
AltiGen IP phone group paging uses SIP Tie-Trunk channels. Make sure that you have a sufficient number of SIP tie-trunk channels configured for group paging.
•
Each paging session requires one G.711 codec channel. The voice stream is multicast to multiple IP phones on the LAN.
•
Any extension (analog or IP) can initiate a paging call by dialing #46 + the Paging Group number.
•
When paged, an IP phone in idle state will automatically turn on the speaker, play a beep, and then play the page.
•
When receiving an incoming call during a paging session, the IP phone will automatically stop the paging session and start ringing.
•
The IP phone user can terminate a paging session by pressing the Release key on the phone.
•
IP phones in DND mode will not be paged.
Some considerations:
•
If an AltiGen IP phone in a different network segment needs to be in a paging group, you need to configure intermediate routers to pass through the IP multicast packets.
•
IP paging to remote IP phones over WAN is not supported.
Note: You can now set up Polycom paging groups; refer to the instructions in the MAXCS Polycom
Configuration Guide.
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To configure paging, select PBX > Paging Group Configuration.
Figure 147. Paging Group Configuration window
Setting Up a Paging Group
1.
In the Paging Group configuration window, below the Group List, click the Add button.
2.
Enter a number for the paging group.
3.
Check the Global Group check box if you want this group to be visible to other gateways, or check
Polycom paging group if this group is for Polycom phones. (Polycom paging groups cannot be Global groups.) Click OK.
Figure 148. Paging group parameters
4.
In the Group Information field, type in the following:
•
First Name and Last Name of the paging group, each with a maximum of 32 characters.
•
Password for the paging group. The default is the system password set on the Number Plan tab of the
System Configuration window.
•
Description for the paging group.
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Adding Members to a Paging Group
Adding Members to a Paging Group
1.
On the Group Member tab of the Paging Group Configuration window, select the desired extension(s) in the Not Member list. Use Shift+click or Ctrl+click to select several extensions from the list.
2.
Click the Add button to move them to the Member list.
Figure 149. Paging Group Configuration, Group Member tab
When a member is added, its default state is Login. Paging group members can use #54 to perform group login or #56 to log off. If a member is logged off, then it will not receive group paging.
Removing Members from a Paging Group
1.
On the Group Member tab of the Paging Group Configuration window, click the extensions that you want to remove in the Member list.
2.
Click the Remove button to move them to the Not Member List.
Polycom Paging Groups
Polycom extensions can make one-way audio announcements to other Polycom users who are in the same
Polycom paging group.
There are now two different sets of Paging Groups in MAXCS:
•
Polycom paging groups (paging group members must have a Polycom Advanced Features license)
•
AltiGen IP Phone paging groups
Polycom Paging Groups can include only users of Polycom phones who have a Polycom Advanced Features license assigned to their extension. Be aware of the following considerations while configuring Polycom Group paging for an extension:
•
Polycom Paging Groups can only page subscribed Polycom phones; if you include a non-Polycom extension in a Polycom Paging Group, the phone on that extension will not play the page.
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•
In order to add an extension to a Polycom Paging Group, the extension must have a registered Polycom phone and it must have a Polycom Advanced Features license assigned to it.
•
A total of 25 Polycom paging groups are available.
•
Each Polycom Paging group must be assigned to a channel ID, 1-25. Channel IDs are shared across multiple MAXCS systems on the same network. In other words, phones subscribed to a Paging group assigned to channel 6 will receive pages for channel 6, no matter which MAXCS system sent the page.
•
Polycom phones allow three paging groups to be specified, Default, Priority, and Emergency. MAXCS
Administrator assigns these three paging groups to channel ID 1, 24, and 25, respectively.
•
The Default group (when enabled) is always shown first in the Paging Group display.
•
A Priority group page will interrupt Normal pages or active calls.
•
An Emergency page will interrupt Normal pages, Priority pages, and active calls and plays out at near maximum volume even if Do Not Disturb (DND) is turned on.
•
Polycom phones must be on the same network in order to send and receive Polycom group pages. If you add an extension to a line park group that is on a different network, then the user will not be able to receive or send pages for that paging group. In addition, if a user relocates and you move the extension to a different network, then that user will no longer be able to receive or send pages to that paging group.
Configure Polycom Paging
Once you have enabled Polycom Paging for an extension, the user can subscribe to paging groups, change the default paging group, configure whether pages play during active calls, and send pages.
To configure Polycom Paging groups,
1.
In MAXCS Administrator, select PBX > Paging Group Configuration.
2.
Click Add to add a new group.
3.
Enter a 3-digit number for the Paging group.
4.
Check Polycom paging group.
5.
Assign a unique paging channel ID number for the group (1-25).
Note: In the future, this channel ID number can only be removed by deleting this Paging group.
Figure 150. Add a new paging group
6.
Click OK. You will see the new group in the Paging Group page, in the left panel. Polycom groups will show (Polycom) to the right of the number.
Modify the group information as needed. For the First Name and Last Name fields, enter details up to a maximum of 32 characters. You can also enter a description for this group.
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Figure 151. Enter information for the new Polycom paging group
7.
To add extensions to Polycom Paging groups, select the group in the list and open the Group Member tab. Move extensions from the Not Member list to the Member list. Click Apply.
8.
Click Reboot affected Polycom phones. This sends the updated configuration to any Polycom extensions that are impacted by the changes; phones with active calls will be rebooted after the call ends.
After the phone reboots, the members of the paging group will see a new softkey on their Polycom phones for that paging group.
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Line Park Configuration
The Line Park feature is a kind of call park method. The main differences between Line Park and system call park are the following:
•
A Line Park ID can be assigned to a specific IP phone's softkey or programmable key; the system call park cannot.
•
Line Park IDs can be grouped as a Line Park Group for call routing purposes; the system call park ID is assigned by the system automatically.
The Line Park feature can be used for the following applications:
•
Inbound call line appearance during business hours
•
Operator parks a call for a group of IP phone users
•
Executive/assistance call coverage
•
Night hours call coverage
•
Overflow new workgroup calls to a Line Park Group when the queue length or queue time is too long.
Implementation notes
•
A total of 99 (01 to 99) line IDs can be grouped into different Line Park Groups. The default "System" group cannot be removed.
•
One Line Park ID can belong to only one group.
•
A Line Park Group can be assigned to:
•
Trunk In-Call Routing
•
Extension/Workgroup Busy or RNA Handling
•
Extension/Workgroup Forwarding
•
Workgroup Quit Queue Option
•
Extensions can be assigned as members of Line Park Groups, allowing the extension users to see and pick up a parked call from those groups in the LinePark tab of their MaxCommunicator or MaxAgent.
•
The system will put the caller in queue when calls exceed the total lines assigned to the Line Park Group.
•
The park line is released when the call disconnects, is answered, or is forwarded due to time out.
Note: You can configure Line Park slots for Polycom phones; refer to the instructions in the MAXCS 8.0
Polycom Configuration Guidel
.
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Configuring Line Park
To configure line park, select PBX > Line Park Configuration.
Figure 152. Line Park Configuration window
Setting Up a Line Park Group
1.
In the Line Park Configuration window, click the Add button below the Groups list.
2.
Enter a name in the dialog box, and click OK.
3.
Select line ID numbers from the Non-Member List and click the Add button to add them to the Member
List.
4.
To assign extensions to a group, select the group, and then click the Configuration button below the
Member Extensions panel.
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The name of the selected group appears here
Figure 153. Configuring a Line Park group’s member extensions
5.
Select members for this Line Park group from the Non-Members list, and click the Add button to move them to the Members list.
Members of a Line Park group can use their MaxCommunicator or MaxAgent applications to see and pick up calls parked for this group.
Any extension can park a call to any group. Any extension can pick up a call from any group using #51 followed by the line park location, if allowed by MaxAdministrator configuration.
6.
Configure the following Line Park options:
Park by System:
•
Play greeting phrase to caller when parked – Select this option to have the system play the greeting phrase you select from the box, before playing music on hold. Specify whether to play the greeting once only, or every x seconds.
•
Play ring back tone to caller when parked – Select this option when you want the caller to hear a ring back tone if the call has not been answered by any extension or voice mail. If the call is answered and parked, the caller will hear a greeting phrase and on-hold music.
•
Enable Timeout – When you check this box, a line park call will time out after the number of seconds set in the value box. Use the Timeout forward to boxes to route the call to an AA, voice mail, or an extension/group.
Park by Extension User:
•
Play greeting phrase to caller when parked – Select this option to have the system play the greeting phrase you select from the box, before playing music on hold. Specify whether to play the greeting once only, or every x seconds.
•
Enable Timeout – Check this box to specify, in seconds, when a line park call will time out. Use the
Timeout option boxes to forward the call to the extension that parked the call, alert the extension that parked the call, or forward the call to an AA, voice mail, or an extension/group.
•
Disable ring tone on IP phone when parked by extension – Check this box to prevent a line-parked call from ringing again while it is parked. (This feature is not supported on Polycom phones.)
Note: The IP phone's programmable key or softkey will be blinking when a call is parked at a line ID that is configured to the phone.
If the associated programmable key has Play Tone function turned on and a ring tone is configured, at the AltiGen IP phone (in idle state) the user will hear a ring tone when a call is parked.
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•
Allow #51 to pick up – when this check box is checked, it allows a user to pick up parked calls from a phone set using #51, followed by the Park Line ID.
Deleting a Line Park Group
1.
In the Line Park Configuration window, select a Line Park Group from the Groups list.
2.
Click the Delete button below the Groups list.
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H A P T E R
21
Workgroup Configuration
The workgroup is an automatic call distribution (ACD) feature designed to enhance customer service operations with queuing, distribution, agent management, real-time status, and call logging capability.
You can configure up to 64 groups, including workgroups, hunt groups, and paging groups.
When adding members to a workgroup, the following rules apply:
•
Concurrent login agent seat license is required.
•
One agent login to multiple workgroups requires only one license.
Creating and Configuring Workgroups
To open the Workgroup Configuration window, select Call Center > Workgroup Configuration.
Figure 154. Workgroup Configuration window, General tab
Overview of Workgroup Configuration Window
These are the tabs in the Workgroup Configuration window:
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•
General – Create workgroup pilot numbers, group descriptions, service level threshold and call recording options.
•
Group Member – Add or remove members from workgroups
•
Business Hours – Set business hours for workgroups
•
Skill Based Routing – Define skill levels and skill-based routing rules.
•
Mail Management – Set capacity and features options for extension mailboxes.
•
Notification – Set preferences and options for voice mail notifications.
•
Call Handling – Set call forwarding, call waiting, and call handling preferences and options.
•
Queue Management – Set queue phrases, overflow routing, queue announcements and queue quit option.
Apply to Button
The Workgroup Configuration window often allows you to apply changes to a particular workgroup or to select many workgroups to which to apply the changes.
Clicking the Apply to button opens a list of all workgroups to which the change can apply. All workgroups are selected by default. You then de-select the ones you don’t want, or de-select all and then select the ones you want. Note that you cannot use the mouse to drag over and select multiple items; you must use the Shift and
Ctrl keys.
The Apply to button is disabled unless there is a change that can be applied to multiple workgroups, and when you use it to apply changes to multiple workgroups, it works on only those changed attributes that can be applied.
Setting Up Workgroups
Set up new workgroups in the Workgroup Configuration window.
To create a workgroup,
1.
Click the Add button under the Group List.
2.
Type in a group number for the workgroup.
3.
Check the Global group check box if you want the group to be visible to other gateways. Click OK.
Establishing Basic Workgroup Attributes
After you create a workgroup, you can set basic attributes on the Workgroup Configuration General tab.
•
First Name and Last Name – Each with a maximum of 32 characters.
•
Password – The default is the system default password set on the Number Plan tab of the System
Configuration window.
A valid password cannot be the same as its workgroup number and must be 4 - 8 digits (numbers or letters
A - Z) in length. Basic password patterns, such as repeated digits (1111), consecutive digit strings (1234), or digits that match the extension (Ext. 101 using 1012, 9101, 10101, etc.) are not recommended. The letters map to numbers (on a phone, for example) as follows:
Numbers
2
3
Letters
A, B, C, a, b, c
D, E, F, d, e, f
Numbers
6
7
Letters
M, N, O, m, n, o
P, Q, R, S, p, q, r, s
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Numbers
4
5
Letters
G, H, I, g, h, i
J, K, L, j, k, l
Numbers
8
9
Letters
T, U, V, t, u, v
W, X, Y, Z, w, x, y, z
•
DID Number – Each workgroup can be assigned a DID number. This number does not have a fixed length, but the length must be long enough (range 2 - 16) for the system to match the DID incoming call.
•
Enable Dial-By-Name Service – Check this box to allow callers to search the list by employee name for this workgroup extension.
•
Description – Describe the purpose of this workgroup.
Setting Call Restrictions
The call restriction rules on the General tab apply to users making outbound calls from within voice mail and several workgroup settings. These settings do not impact the call restriction settings configured for the workgroup member's extension in Extension Configuration.
•
Allow Calls to be Transferred or Conferenced to an Outside Number – When checked, the internal extension user can log into this workgroup voice mail, make a call to a second party, then transfer or conference to a third party.
•
Allow User to Configure Forwarding, Notification, and Reminder Call to an Outside Number – This setting regulates workgroup call forwarding, voice mail notification, and reminder call configuration. If this setting is not checked, you will see a warning message pop up when trying to set up forwarding to an outside number. International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Return Calls from within Group's VM System – When checked, an outside caller can dial into the system, log in to workgroup voice mail, and make or return calls from the group's voice mail (Zoomerang feature). International calls are not allowed if the fourth option is not checked.
•
Allow Outside Caller to Make or Forward International Calls from within the Group's VM system –
This setting regulates making international calls from voice mail and forwarding to an international number.
Caution!
Allowing any of these options may increase the potential for toll fraud. Make sure the password is properly configured to prevent an intruder from using this voice mail box to make an outbound call.
AltiGen recommends that you leave the fourth option unchecked for all workgroups at all times.
Service Level Threshold
The Service Level Threshold scroll box allows you to select the length of time in seconds that a call can be in queue before the call is logged in workgroup performance statistics as having exceeded the allowable service level limits. You can set the value to any number between 1 - 1200 seconds.
Service level is a service quality index which calculates the percentage of calls serviced within a defined threshold for the defined period of time. The term "serviced" may not necessarily mean answered. You can define the calculation method based on your operation requirements. The service level percentage is calculated from midnight 00:00 am. and is reset daily. The calculated number will be output to the MaxAgent and Max-
Supervisor applications.
The Service Level Calculations Options button opens the following dialog box.
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Figure 155. Service Level Calculation dialog box
In the Method Options section, select one of the following:
•
TSL (Total Call Service Level) – The service level calculation is: TSL% = Total WG inbound calls within
SLT / Total WG inbound calls. This is the default option.
•
ASL (Answered Service Level) – The service level calculation is: ASL% = Total WG inbound calls answered within SLT / Total WG inbound calls.
•
Customize – Use the check boxes to enable at least one of the following three options:
–
Answered Calls within Service Level
–
Abandoned Calls within Service Level
–
Overflowed/Redirected Calls within Service level divided by at least one of the following three options:
–
Total Answered Calls
–
Total Abandoned Calls
–
Total Overflowed/Redirected Calls
Workgroup Recording Options
The system administrator can specify the following workgroup call recording options for a workgroup:
Warning! Listening in to or recording a conversation without the consent of one or both parties may be a violation of local, state and federal privacy laws. It is the responsibility of the users of this feature to assure they are in compliance with all applicable laws.
•
Disable – No call recording.
•
Auto record to central location – Records all workgroup inbound and outbound calls, which are saved
to a central location (defined in Recording Configuration on the System menu – see Call Recording
); this option requires that either a shared Concurrent Recording Session license is available or that a Dedicated Recording Seat license is assigned to each workgroup member (configured in
Extension Configuration).
•
Record on demand to central location – Records calls on demand, which are saved to a central location
(defined in Recording Configuration on the System menu – see see Call Recording Configuration); this
option requires that either a shared Concurrent Recording Session license is available or that a dedicated
Recording Seat license is assigned to each workgroup member (configured in Extension Configuration).
•
Record on demand to extension VM – Records calls on demand, which are saved to the agent’s voicemail box.
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Note: When retrieving voice mail as an e-mail, if the voice mail file has a recorded file attached, the recorded file is not forwarded in the e-mail.
•
Insert Recording Tone – Plays a recording beep to alert the parties that the conversation is being recorded, then plays a periodic recording alert tone. The tone is recorded together with the conversation.
•
Record X out of 10 calls – If recording to a central location, automatically records incoming and outgoing
workgroup
calls, as specified. (The default is to record all workgroup calls.)
To see this option, click the Agent Recording Management button.
You can change these values
For each agent you can change the option Record N out of 10 calls. For example, if you set to record 4 out of 10 calls, the 1st-4th and 11th-14th, and so on, will be recorded. Using this example, in the following table the shaded calls will be recorded:
To change Record N out of 10 calls for an agent, click the cell you want to change, and make a selection from the list. Click Apply. When finished, click OK.
•
Centralized Recording – You can also enable or disable centralized recording from the Agent
Management Recording window shown above. Click the cell you want to change, and make a selection from the list. Click Apply. When finished, click OK.
Notes:
•
The recording session starts when the call enters the connected state and ends when hang up or flash is pressed, or when the call is transferred.
•
The recording setting at Extension Configuration applies only to non-workgroup calls. The recording setting at Workgroup Configuration applies only to workgroup calls. To allow an agent to record all calls (non-workgroup and workgroup), both recording settings must be enabled.
•
When an agent logs in to a workgroup, which is also an outbound workgroup, all outbound calls will be considered as workgroup calls and recorded according to workgroup configuration.
•
When an agent logs in to a workgroup and is in Not Ready, DND, Wrap-up, or Inter-call Delay state, outbound calls will be recorded if workgroup recording is configured.
•
When an agent does not log in to the workgroup that is configured as an outbound workgroup, all outbound calls are non-workgroup calls.
Recording Tone Options
When you select to record conversations, you can have the system insert a repeating recording tone.
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On the General tab of the Workgroup Configuration window, set Recording Tone to Insert repeating
recording tone.
With MAXCS systems, the repeating tone is only available if one channels in use is SIP. With two TDM channels, there can be no repeating tones.
Figure 156. The Repeating recording tone option
Establishing Workgroup Membership
Add agent extensions to a workgroup on the Group Member tab in the Workgroup Configuration window.
Figure 157. Workgroup Configuration, Group Member tab
To add extension(s) to a workgroup,
1.
Select the workgroup in the Group List.
2.
On the Group Member tab, click the extension number(s) in the Not Member list. Use Shift+click and
Ctrl+click to select several extensions.
3.
Click the Add button between the columns to move them to the Member list.
Note: If the workgroup pilot extension is configured to Ring All Available Members, the maximum number
of members is 20. See “Setting Call Handling Options” on page 277 for details.
By default, a newly added member has the Skill Level set to 1.
4.
To change the Skill Level designation for a member, double-click the member in the Member List. The
Skill Level dialog box opens. (Skill Levels are defined in “Skill Based Routing” on page 268.)
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Setting Business Hours
Figure 158. Skill Level dialog box
5.
Click the desired Skill Level Index, then click OK.
Agents who are members of more than one workgroup can have a different skill level assigned in each group.
To remove extension(s) from a workgroup,
1.
Click the extension number(s) in the Member list.
2.
Click Remove to move them to the Not Member list.
Log In/Out a Group Member
An administrator can log in or log out a group member, by selecting the member in the Member List and clicking the Login Now or Logout Now button.
Setting Login Status for System Restart
Whenever the system is restarted, the administrator can use the list at the bottom of the Group Member tab to:
•
Keep Login Status – All group members retain their original login status for that group prior to restart
(default setting)
•
All Login – All group members are automatically logged into the assigned group after the system is restarted.
•
All Logout – All group members are logged out of the workgroup when the system is restarted.
Setting Business Hours
Settings on the Business Hours tab in the Workgroup Configuration window define how after-hours calls are handled for workgroups. An administrator can assign a Business Hours profile to a group, and also configure after-hours handling for each day of the week.
To set after-hours call handling, select the workgroup you want to work with from the Group List in the
Workgroup Configuration window, then click the Business Hours tab.
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Figure 159. Workgroup Configuration, Business Hours tab
Set the business schedule parameters as follows:
Workgroup
Business hours
Parameter
Description
Business Hour
After Hours/Non-
Workday Handling
Logout All Agents At
Use the list to select a Business Hours profile to apply to the workgroup
(profiles are configured in the System Configuration window - see “Setting
•
For each day of the week, select a Forward To option for call handling after hours or for non-workdays:
•
To Voice Mail
•
To AA – Select the auto attendant to use in the list under the option.
AAs are configured in the AA Configuration window, available from the
System menu.
•
To an Extension – Select an extension from the list.
•
To a Group – Select a group from the list.
•
To the Operator
•
To an Outside Number – If you choose Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
•
To an App Ext – When used in conjunction with a third party notification application, the App Ext feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log- on extension to which the third party application is connected.
Contact your local AltiGen Partner for more information on using this feature.
For each day of the week, you can select up to three time periods for the system to automatically log out agents.
Skill Based Routing
If you want to set up skill-based routing, you can more closely match a customer’s call to an agent who has the skills needed to handle that customer’s issue.
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Skill-based routing can increase customer issue resolution on the first call, lower the abandoned call rate, and in turn increase customer satisfaction.
The Skill Based Routing tab in the Workgroup Configuration window lets you define up to nine different levels of skill needed to handle the variety of a workgroup’s calls.
Skill definitions
Agents who are assigned to the selected skill number
Figure 160. Workgroup Configuration, Skill Based Routing tab
Skill number 1 could define the most basic skill and level 9 the most advanced, or vice versa. Or the skill numbers can be used in any other way that works for the way your company does business.
After skill numbers have been defined on this tab, each agent in the workgroup should be assigned a skill number, according to that agent’s knowledge and ability, on the Group Member tab.
Incoming calls can be set to ring agents according to skill number, thus more closely directing the caller to an available agent qualified enough to help the caller, but ideally not over-qualified. You can determine the skill required by the caller and set the SKLR number in several places:
•
The auto attendant, depending on the caller’s responses (see “Configuring Menu Items” on page 82)
•
The DNIS number the caller dialed, depending on how you have set up your DNIS numbers (see “Defining
•
The caller ID (see “Defining Caller ID Routing” on page 164 )
•
The Advanced Call Router – You can define SKLR in each rule entry in the Call Router, and if the Call
Router routes a call, SKLR will be set.
•
In the SDK – A call's SKLR can be set in some modules, and now, we just support APC interface, that is if a call is connected to an App Ext, this App Ext can set or change the call's SKLR
You can set rules on the Skill Based Routing tab to allow all calls coming into a workgroup to be handled by agents with a lower skill number or a higher skill number than is set for a call. And you can set time-based rules that alter the call’s SKLR to allow either less able agents or over-qualified agents to handle a call so that the caller does not have to wait for an excessive period of time.
Note: For the settings configured on the Skill Based Routing tab to take effect, you must select the Skill-
Based Routing option on the Call Handling tab of the Workgroup Configuration window (see “Setting
IntraGroup Call Distribution” on page 279
).
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Operational Limitation
When configuration Skill based routing, be aware of a known operational limitation with queue announcements. If the announcement is shorter than the escalation time, then the system escalates the call at the end of the announcement instead of waiting for the escalation time to expire.
Defining Skills for a Workgroup
1.
Select a workgroup in the Group List.
2.
Double-click a skill number in the Skill Definition list, or select a skill number and click the Edit button.
3.
In the Skill Level Name dialog box, enter the skill name in the Description field, then click OK.
Figure 161. Skill Level Name dialog box
The description appears in the Skill Definition list for that skill number.
Setting Rules for Skill Based Routing
The Skill Coverage Rule on the Skill Based Routing tab establishes the pool of agents who can handle a particular workgroup call, based on the SKLR setting for that call. The group may comprise:
•
Only agents assigned that skill number
•
Agents with a given skill number and lower
•
Agents with a given skill number and higher
This setting must be configured.
To further help ensure that a workgroup is handling calls in a timely manner, you can specify how many seconds a caller can be in queue before opening the call to agents with the next skill number up or the next skill number down, in successive steps.
To set skill-based routing rules,
1.
In the Workgroup Configuration window, Skill Based Routing tab, select the workgroup for which you want to set the rules.
2.
Select an Agent’s Coverage Rule
•
Exact Match SKLR of Incoming Call
Only agents whose skill number matches the SKLR of the incoming call can answer the call. For example, if you have three callers with SKLR equal to 2 in the workgroup queue, and all agents with skill level 2 are busy, and there are agents with skill level 1 and 3 who are idle, the system will keep the callers in queue waiting for an agent with skill level 2 to be available.
•
Equal or Lower than SKLR of Incoming Call
Any agent whose skill number is equal to or lower than the SKLR of the incoming call may handle this call. Agents with the lowest skill number are rung first. With this option, that would be agents whose
es you are willing to use for this type of call. For example, you can set a regular call's SKLR to 1 and a preferred customer's SKLR to 3. Calls from preferred customers can be answered by agents with skill level 3, 2, and 1 while regular calls can only be answered by agents with skill level 1.
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•
Equal or Higher than SKLR of Incoming Call
Any agent whose skill number is equal to or higher than the SKLR of the incoming call may handle this call. Agents with the lowest skill number are rung first. With this option, that would be agents whose
skill number matches the SKLR. Set the SKLR (see “Skill Based Routing” on page 268) as if you were
setting a minimum skill level requirement for the call. For example, say a technical support group has agents with skill level 1 (beginner), 2 (intermediate), and 3 (expert). If you select the "Equal or Higher" option, calls with SKLR 2 will be queued for an agent with skill level 2 or 3.
3.
To increase coverage of calls, check the Enable SKLR Expansion check box. (This check box is available if you selected the Equal or Lower option or the Equal or Higher option.)
4.
For each level, specify the number of seconds a call can be in queue before the system will include the next level of agents in the pool of agents who may handle the call. Either use the Up/Down arrows or type in a number from 1-999.
Skill Based Routing Examples
Example 1: Coverage rule is Equal or Lower and Enable SKLR Escalation is checked.
The above configuration means:
1.
When a caller with SKLR 1 is waiting in queue for 30 seconds, the caller's SKLR will be escalated to 2.
Agents with skill levels 1 and 2 are able to handle the call.
2.
If the caller stays in queue for more than 60 seconds, the caller's SKLR will be escalated to 3. Agents with skill levels 1, 2, and 3 are able to handle the call.
3.
If the caller stays in queue for more than 90 seconds, the caller's SKLR will be escalated to 9 because all other escalation wait times are set to 0 seconds. The call will be distributed any idle agent in the workgroup.
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Example 2: Coverage rule is Equal or Higher and Enable SKLR Escalation is checked.
The above configuration means:
1.
When a caller with SKLR 9 waiting is in queue for 30 seconds, the caller's SKLR will be changed to 8.
Agents with skill level 8 and 9 are able to handle the call.
2.
If the caller stays in queue for more than 60 seconds, the caller's SKLR will be changed to 7. Agents with skill level 7, 8, and 9 are able to handle the call.
3.
If the caller stays in queue for more than 90 seconds, the caller's SKLR will be escalated to 1 because all other escalation wait times are set to 0 seconds. The call will be distributed to any idle agent in the workgroup.
Setting Workgroup Mail Management
The Mail Management settings define how voice messages are handled for a workgroup, including how messages are announced and processed, and how much capacity is allotted to message storage.
To work with mail management settings, click the Mail Management tab, and select the workgroup number you want to work with from the Group List.
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Setting Workgroup Mail Management
Figure 162. Workgroup Configuration, Mail Management tab
Note: You can use the Apply to option to apply mailbox settings to one, some, or all workgroup.
Disabling a Mailbox
When you disable a mailbox, the normal greeting is played but callers cannot leave messages.
Setting E-mail Options
On the Mail Management tab, you can set the e-mail options for the workgroup:
•
E-mail Name – The workgroup’s e-mail name without the @domain. The default e-mail name is ext
<workgroup number>,
that is, the letters “ext” followed by the workgroup number. For example, the default e-mail name for workgroup 500 would be ext500.
•
Retrieve Voice Mail by E-mail Client – When selected, this sends voice mail to the user extension as an e-mail attachment. Deselected, voice mail is retrieved as voice mail.
•
Enable Mail Forwarding – When selected, the workgroup’s e-mail will be forwarded to the e-mail address you specify in the Forward E-mail Address box. The address should be a full address, including the domain (for example, [email protected]).
If you enable mail forwarding, you also specify what you want done with the original messages after they have been forwarded. In the list you can choose to:
•
Delete Messages after Forward
•
Keep the Messages as New
•
Keep Messages as Saved
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Setting Mailbox Playback Options
You can use the following check boxes to turn on or off options for listening to playback of recorded messages.
These options apply to both new messages and saved messages, and they can be applied to multiple workgroups using Apply to:
Message Playback
Parameter
Announce Message Sender
Before Playback
Announce Time Stamp
Before Playback
Confirm Callback Number
Enable Distinctive Call
Waiting Tone
Play the Newest Voice
Message First
Description
Selected, the user hears the name of the message sender (internal sender only) before listening to recorded AltiGen Voice Mail System messages.
Selected, the user hears the timestamp (time and date) of each message before playback.
Selected, system confirms the accuracy of the caller’s number.
Selected, the user hears three different call waiting tone cadences to
distinguish between internal, external, and operator calls (see “Distinctive
Selected, new voice mail will be retrieved first. When not selected, the system will play voice mail based on FIFO (first in, first out).
Setting Mailbox Capacities
You can set various mailbox capacities with the following options, and you can apply the settings to multiple workgroups using Apply to:
Mailbox Capacity
Parameter
Max Number of Messages
Mailbox Size
Max Message Length
Retention Length of Saved
Description
Maximum number of messages stored in the workgroup’s mailbox. The range is 1 - 999, defaulting to 100.
Mailbox size in MBs of stored messages. The range is 1–500 MB, with a default of 50.
Maximum length of voice messages in minutes. The range is 1–30 minutes, with a default of 5 minutes.
Number of days saved messages are archived by the system. The range is
1–90 days, with a default of 60.
Press Zero Option
This option allows a caller to press “0” while listening to this workgroup’s greeting. When the caller presses “0,” the call will forward to the specified destination. Use the list to specify a forwarding destination for the call:
Voice Mail, AA, Extension, Group, Operator (default), Outside Number, or Line Park.
If you choose to forward to an Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
Voice Mail Access Option
To allow agents of a workgroup to access the group’s voice mail in MaxAgent (MaxAgent’s WG VM tab), select the group and check Enable agents to access voice mailbox of workgroup.
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Setting Message Notification Options
Setting Message Notification Options
To set notification options on new incoming e-mail and voice messages, click the Notification tab in the
Workgroup Configuration window, and select the workgroup number from the Group List.
Figure 163. Workgroup Configuration, Notification tab
Individual users can also configure Message Notification within the AltiGen Voice Mail System.
Note: You can use the Apply to option to apply mailbox settings to one, some, or all workgroups.
Setting the Message Types for Notification
Select the types of messages for which the workgroup user will be notified:
•
None – When selected, the user is not notified with a call regarding newly received messages. Selecting this option does not prevent the user from getting message waiting indicators or stutter dial tone when new messages are received.
•
Urgent Voice Messages Only
•
All Voice Messages
Please note that the system will start notification as soon as it enters non-business hours under the following conditions:
•
Extension is set to notify Urgent Voice Message Only
•
Notification is set to Non-Business Hours
•
Voice mail is received during business hours and is marked urgent
•
Extension user does not check the urgent message
Setting the Type of Notification
There are several options for sending the notification or reminder message: phone, pager, extension or
custom application (Custom App).
•
Extension – To use the Extension option, select the Extension radio button, then type the extension number into the text box.
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Phone/Pager – For the Phone and Pager options, first specify the trunk or route access code using the list next to the Phone radio button. The Any option means to locate any available trunk. Then type in the number with all relevant dialing prefixes other than the trunk code, using a maximum of 63 digits.
•
Custom App – When used in conjunction with a third-party notification application, the Custom App feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log-on extension to which the third-party application is connected. Contact your local
AltiGen Partner for more information on using this feature.
Note: The Reminder Call will not work with this selection.
Note also the following considerations:
•
For the Pager option, the system calls the specified pager number and then dials the system main number
(as set in System Configuration, General tab), which is then displayed on the user’s pager.
For the operator-assisted paging function, the operator phone number and the pager number must be entered in the <phone number>*<pager number> format. For example, if the phone number to call the pager operator is 7654321 and the pager number to page the user is 12345678, the notification outcall number that needs to be entered is 7654321*12345678. When the pager operator answers the Message Notification call, MAXCS announces the pager number and the System Main Number (as configured on the General tab of System Configuration), which will be displayed on the user’s pager. The operator is also given the option to repeat these numbers by pressing ‘#’.
Outcall to Cellular or PCS Phone Numbers
When an outcall is made by the system (for One Number Access, Message Notification, Zoomerang, Call Forwarding, and so on) to a cellular or PCS phone, it may ring the phone once but not necessarily present the call and make a connection. This will happen if the ringback tone played by the cellular service provider does not conform to standard ringback tones. To work around this problem, append a few commas (,) to the outcall (cellular) number when entering it. Each comma provides a one second pause.
Setting Notification Timing
When notification is configured to an outside phone number, the system will announce, "This is the outcall notification message for…" after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the carrier. If the system plays the announcement phrase before the notification call is answered, the phrase will be cut off. The following two options can be configured based on answer supervision capability:
•
Seconds after Dialing – If the carrier of the outside phone number cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
Note: If the delay is set too long, the notified party will hear silence before the announcement is played.
•
Seconds after Answered – This field is set to 0 seconds and it is not configurable for notification to a phone number. It means the system will play the announcement immediately after answer supervision is received.
When notification is configured to a pager, the system will transmit DTMF digits as the return phone number
(the System Main Number as set in the System Configuration General tab) after call connection. However, there are situations when the system may not be able to receive an answer supervision signal from the pager system. If the system sends digits before the call is connected, some digits will be cut off. The following two options can be configured based on answer supervision capability:
•
Seconds after Dialing – If the pager carrier cannot provide an answer supervision signal, check this option and set a delay time. (Default 5 seconds, maximum 30 seconds.)
•
Seconds after Answered – If the answer supervision signal is provided by the carrier, check this option and set the delay timer to 2 to 5 seconds. In some cases, the pager carrier cannot detect DTMF right after the call connection. (Default is 10 seconds, maximum is 30.)
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Note: You may need to try a different delay setting to make sure the user return number is transmitted properly after configuration.
Setting Notification Business Hours
You can choose one of three options for when the extension user is to be notified of new messages:
•
Non-Business Hours – Notify only during non-business hours. Business hours are set in System
Configuration, Business Hours tab (see “Setting Business Hours” on page 56).
•
From/To – Notify during a specified time of day. Select the hours in the From and To time scroll boxes.
•
Any Time – Notify at all times (every day).
Setting Call Handling Options
Call Handling options include forwarding, handling busy calls, handling no-answers and other options.
You can use the Apply to option to apply call restriction settings to one, some, or all workgroups.
To work with workgroup call handling options, click the Call Handling tab in the Workgroup Configuration window, and select the workgroup number from the Group List.
Figure 164. Workgroup Configuration, Call Handling tab
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Handling Busy Calls
You have several options for handling calls when the workgroup extension is busy. If you do not enable busy call handling, the caller simply hears a busy signal.
To enable the options, select the Enable Busy Call Handling check box, then select from the following forwarding options:
•
Group Queue – The caller will stay in the workgroup queue waiting for any agent to become available. If there is no agent logged in at this moment, the system will use Group Logout Handling to handle this call.
•
Group Voice Mail
•
AA – Forward caller to an auto attendant.
•
Extension – Forward caller to an extension.
•
Group – Forward caller to another group.
•
Line Park – Forward caller to a Line Park group.
Forwarding All Calls
When you do not want the workgroup to handle any calls, check the Enable Forward To option in the Forward
All Calls section of the Call Handling tab, and select an option.
The forwarding options are as follows:
•
To Voice Mail
•
To an Extension – Select an extension number in the drop-down list.
•
To AA – Select the AA to use in the list below the option.
•
To a Group – Select a group from the list.
•
To the Operator
•
To an Outside Number – This option is available if it is allowed in the Other Call Restrictions option in
Cellular or PCS Phone Numbers” on page 276.
If you choose Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
•
To an App Ext – When used in conjunction with a third-party notification application, the App Ext feature enables an extension to connect to an application that can receive the notification event; use the list to choose the log-on extension to which the third-party application is connected. Contact your local AltiGen
Partner for more information on using this feature.
•
To Line Park – If configured, select a Line Park group from the list.
Handling Unanswered Calls
The Enable No Answer Handling configuration provides options for handling calls when the system rings the first available agent and the call is not answered. If all agents in the workgroup are rung and no one answers the call, the system will use the Group RNA/Logout Handling rule. Enable No Answer Handling is not available if Intra Group Call Distribution is set to Ring All Available Members.
To configure this option, check the Enable No Answer Handling box.
Select one of the following forwarding options for no answer call handling:
•
Next Group Member – Ring the next available agent until all available agents are rung. If all agents are busy, caller will stay in the workgroup queue.
•
Extension – Take the call out of the workgroup and forward it to an extension.
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•
Group – Take the call out of workgroup and forward it to another group.
•
Group Voice Mail – Transfer the caller to the workgroup voice mail when the first available agent does not answer the call.
•
Member Voice Mail – Transfer the caller to the first available agent's voice mail if this agent does not answer the call.
•
AA – Take the call out of the workgroup and forward it to an auto attendant.
•
Line Park – Take the call out of the workgroup and forward it to a Line Park group.
Set RNA Agent Auto Logout Check Box
Select this option to have the system automatically log out an agent extension from a workgroup if RNA is encountered.
Set RNA Agent Not Ready Check Box
Select this option to have the system automatically set an agent’s extension in a workgroup to not ready if RNA is encountered.
Number of Rings Before Handling
If you select Ring All Available Members in the Intra Group Call Distribution section, then specify the Number
of Rings before Handling, using the scroll box beside that option. The number of rings is the total number of times agents are rung before the call is handled by the Group RNA/Logout Handling configuration
Setting IntraGroup Call Distribution
The IntraGroup Call Distribution options let you set the handling of normal inbound calls: how to route the incoming call to a workgroup agent, using one of the following options:
Figure 165. The IntraGroup Call Distribution options
•
Ring First Available Member – First available extension in a workgroup. For example, if there are three member extensions in a workgroup, the call is always sent to the first member configured in the workgroup. If this member is busy, the call goes to the second member configured and so forth.
•
Ring Next Available Member – Around-robin method that attempts to evenly distribute calls among the group members. This method sends the call to the next member configured in a workgroup (regardless of whether the previous member is busy or not). In other words, if the previous call was sent to #3 in the group, the present call is sent to #4, if #4 is not busy.
•
Ring All Available Members – All extensions in a workgroup.
Note: When this option is enabled, a single workgroup can have no more than 20 members.
In addition, calls to the workgroup with this option enabled have higher priority than other workgroup calls. Therefore, if an agent belongs to multiple workgroups, one of which has this option enabled, a call to that workgroup will be processed first, regardless of Wait Time of calls in other workgroups which are not set to Ring All.
If members are using IP extensions, the system will not use the IP codec channel during ringing all IP
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•
Ring Longest Idle Member – The agent who has the longest idle time, defined as follows:
•
The agent needs to be in login state
•
Idle time is calculated from the end of the last wrap-up event.
•
If the agent does not have wrap-up time configured, the idle time is calculated from the end of last busy state.
•
Ring Average Longest Idle Member – The agent who averages being off the phone the longest:
•
Average Idle Time = (Total Login Time – Total Talk Time) / Total Calls Handled
•
Total Login Time = Cumulative WG login time since midnight
•
Total Talk Time = Cumulative WG Inbound + Outbound call duration since midnight
•
Total Calls Handled = Total number of WG calls (Inbound + Outbound) handled by the agent since midnight
If a new agent logs into a WG that has been operating for several hours, this agent will have the highest priority to take the call.
If multiple agents log into a WG that has been operating for several hours, the one with the longest idle time since login will have the highest priority to take the call.
•
Ring Fewest Answered Calls – The agent who has answered the fewest number of calls.
•
Ring Shortest Average Talk Time – The agent who averages the shortest talk time.
Note: Average talk time is calculated as follows:
Average Talk Time (ATT) = Total Inbound Talk Time / Total Inbound Calls
Answered
The agent calculated with the lowest value for ATT is rung first.
•
Skill-Based Routing – The call will be routed according to the SKLR setting and skill-based routing rules
take the call when multiple agents with the same skill level are idle.
Enable Single Call Handling for Agents
Check this check box to enable single call handling for workgroup agents.
•
If single call handling is enabled and the agent has one or more calls on hold, MAXCS will not distribute the call to this agent. If single call handling is disabled, MAXCS will distribute calls to this agent even when one or more calls are put on hold by this agent.
•
If a workgroup agent uses a Polycom phone, Single Call Handling will be automatically enabled, regardless of the workgroup’s Enable Single Call Handling for Agents setting.
Handling Calls when Group Members Are RNA/Logged Out
You can set calls to forward to a specified destination when all group members either do not answer the call
(RNA) or are logged out. To do so, in the Group RNA/Logout Handling section of the Call Handling tab, check the Enable Forward to check box, and select a destination from the list.
The forwarding options are described in “Forwarding All Calls” on page 278. An additional option,
Group Queue
, forwards incoming workgroup calls to the workgroup call queue. With this setting, when agents log back in, they can begin accepting the queued calls
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Announce Agent Info
Check this check box to have the system announce the agent’s directory name before an incoming workgroup call is transferred to an agent from the queue.
Inter Workgroup Call Distribution
In the case where an agent belongs to multiple workgroups and there are queued calls in two or more of these workgroups, as soon as the agent becomes available, the queued call that will be distributed to this agent is determined by the Inter Workgroup Call Distribution setting.
This field is used to calculate the score of each call in a workgroup’s queue. Depending on the call’s assigned priority and the skill of the agent that is available, the score will determine which workgroup’s call gets answered first. The call with the highest score is answered first. Use the up/down arrows to increase or decrease the weight values for Priority and Skill values.
The first box is the weight for agent skill in a workgroup. The second box is the weight for priority of a queued call. The score is calculated as (10 – skill level) x weight for agent skill + (10 – queued call priority) x weight for call priority. When determining which call should be dispatched to an idle agent who is assigned to multiple workgroups, the system will consider the following factors:
•
Caller's priority
•
Agent's skill level
•
Caller's SKLR
•
Caller's wait time in queue
Configuration Guidelines:
•
Assuming an agent is assigned with different skill levels for different workgroups, and call priority is the same for all calls, you can increase the skill weight to 9 and reduce the priority weight to 1 to better match an agent's skill.
•
Assuming each call is assigned with a priority based on certain customer attributes, and an agent's skill is the same for all workgroups, you can increase the priority weight to 9 and lower the skill weight to 1 to have a call with higher priority answered first.
•
Assuming all calls' priority is the same and agent's skill level is the same for all workgroups, you can use this scoring system to prioritize workgroups. For example, assign priority weight 9 to the most important group, 5 to the second most important group, and 1 to the least important group. Calls in the group with higher priority weight will be dispatched first.
•
When there are callers with the same score in different workgroups, the queue time will be used as a tie breaker.
•
If you have variable priority settings for callers, and agents belong to multiple workgroups with different skill levels, it is recommend that you set all calls' SKLR to 1 and set call coverage rule to "Equal or Lower than SKLR of Incoming Call". This will eliminate the complexity of matching caller's SKLR to agent's skill level.
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Queue Management – Basic
The main differences between Basic and Advanced queue control are the following:
•
You can build a layer of DTMF menu selection option in the Advanced queue control mode. A caller can press a digit to hear different prompts or options while in queue.
•
You can have select FIFO (First In First Out) or LIFO (Last In First Out) for the queue overflow.
If you don't have these particular needs, the Basic queue mode is recommended.
The Queue Management tab in Workgroup Configuration allows you to set options for queue phrases and announcements, queue overflow routing and quit queue options. Options become enabled depending on the
Queuing Control selected – Basic, Advanced and Application Extension.
Figure 166. Workgroup Configuration, Queue Management tab, Basic Queue Control
When Basic is selected in the Queuing Control field, the following queue management options are available:
Setting Queue Phrase Options
For each workgroup, you can either use the system default phrases or you can set up a custom configuration.
The default audio phrases are discussed in “Audio Peripheral Configuration” on page 67.
Queue Announcement
You can set up the system to announce a caller’s queue status – queue position and expected queue time – when an incoming call enters a workgroup queue. To enable this option, check Enable Announcement, then check Queue Position and/or Expected Queue Time.
Queue Position – When checked, the system will tell the caller which position the caller is at in queue. Do not check this option if you assign different priorities to different calls based on DNIS, CID, or AA selection. Do not check this option if you configure matching a caller’s SKLR to an agent's skill level. Queue position is not meaningful when a higher priority caller can push a lower priority caller or if no agent is available to answer a particular SKLR.
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Expected Queue Time – when checked, the system will tell the caller how long the wait is expected to be.
When calculating this number, the system will consider the average agent call handling time and the position of the caller in queue. Because queue position is a factor when calculating this number, do not check this option when call priority and caller SKLR matching are configured. Please note that the Expected Queue Time is an estimated number. Agents logging in or out of the workgroup during operation hours will affect the actual handling time and cause deviation to the expected queue time.
When Avg time in queue is less than 1 minute, the expected wait time in queue will not be played to the caller.
Expected Queue Time
(round up to minutes)
= [(Average Call Handling Time x Queue
Position) + 59 sec] / 60 sec
Expected Wait Time Sampling
To calculate Expected Queue Time, the system needs to take samples when a workgroup starts operation.
You can set the following parameters to set a sampling period and a fixed Expected Queue Time announcement during sampling period. The expected queue time counter is reset for all workgroups daily at midnight.
•
Initial Sample Call Count [1 to 100] – How many calls you would like to use as initial samples.
•
Initial Expected Wait (Queue) Time [1 to 10 minutes] – This field defines the expected queue time to be announced during the sampling period. One minute is the minimum you can set for expected wait time in queue.
Queue Overflow Forwarding
The Queue Overflow Forwarding options are for handling long queues or long wait times for callers. When a queue exceeds a set number of calls, or callers are waiting beyond a set length of time, calls can be automatically forwarded to a voicemail box, AA, extension, group, operator, outside number, or application extension.
To set options for handling queue overflow,
1.
In the Queue Overflow Forwarding section, set options for:
•
Calls in queue exceed – When the number of calls in queue are greater than the defined number, new incoming calls will be overflowed to the defined target.
•
Expected queue time longer than – When the longest queue time is greater than the specified number of minutes, new incoming calls will be overflowed to the defined target.
•
Service level lower than – This number is not a historical service level defined in the workgroup threshold. This number is a real-time queue service level (RTSL) calculated as follows:
RTSL% = 1 – (# of queued calls exceed SL threshold / Total calls in queue)
2.
Check the Enable Forward to check box and from the list, select the forwarding destination list to use if the queue length, wait time or service level settings are exceeded. If this option is not checked, calls will go to the workgroup’s voicemail.
Quit Queue Option
The quit queue feature gives a caller the option of leaving a workgroup queue at any time by pressing # and/ or 0. To enable this feature, check either or both of the Enable Quit Queue Options, then use the appropriate
Forward to list to select the option the caller will have:
•
Voice Mail – Sends to voicemail
•
AA – Select the auto attendant to use. AAs are configured in AA Configuration on the System menu.
•
Extension – Select an extension from the list.
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Note: If the forwarding extension is busy when a caller quits a queue, the call will go to this extension’s voice mail.
•
Group – Select a workgroup from the list.
•
Operator – Sends to the operator
•
Outside Number – This option is available if it is allowed in the Other Call Restrictions option in the
Restriction tab, as discussed in “Setting Other Call Restrictions” on page 198.
If you choose Outside Number, select a trunk or route access code to use in the small list on the left, and type in the full prefix and phone number.
Note: Forwarding calls to a pager is possible but not recommended since callers will only hear what is heard when calling a pager and will not know to enter a return phone number unless instructed.
•
App Ext – When used in conjunction with a third-party notification application, this feature enables an extension to connect to an application that can receive the notification event; use the list to choose the logon extension to which the third-party application is connected. Contact your local AltiGen Partner for more information on using this feature.
•
Callback Interview – The System will record the caller’s callback number and will prompt the caller to record a message into the voice mail box of the workgroup.
Note: This option is only available to external callers.
Priority Promotion
To prevent calls with lower priority staying in queue forever, causing high abandon rate, or lowering service level, you can set priority promotion to enhance a caller's position in queue. Check the box and enter the proper time interval in seconds.
Supervisor Queue Control
When the Allow Redirect Call/Change Priority check box is checked, this allows a workgroup supervisor to redirect queue calls or change the call priority of queued calls, using the MaxSupervisor application.
Queue Management – Advanced
When Advanced is selected on the Queue Management tab, the Setup button becomes available.
To configure options for advanced queuing control, click the Setup button. This opens the Advanced Queue
Management (AQM) application configuration window with tabs for configuring Announcement, Menu
Selection, Queue Overflow, and Callback from Queue.
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Figure 167. Workgroup Configuration, Queue Management, Advanced Queue Control
Operational Notes
•
If your system has a lot of extensions, there may be a slight delay opening the Advanced Queue panel.
•
Once you configure advanced queue control settings for a workgroup, by default, callers to that workgroup can no longer press the # key to leave a voicemail message. If your queue announcement tells callers that they can press # to leave a message, consider updating the appropriate phrase.
Announcement Tab
The Announcement tab allows for configuration of queue announcements.
Figure 168. Workgroup Configuration, Advanced Queue Management, Announcement tab
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To configure queue announcements:
1.
Select any of the following check boxes:
•
Use Default System Phrases
•
Queue Position
•
Expected Wait Time
2.
If you are not using default system phrases, use the lists to select the Greeting Phrase and Update
Phrases that will be played to callers in queue.
3.
Select the Update Interval (0 - 180 seconds) to be inserted between queue phrases.
Note: If the interval is set to 0, the system will play phrases one after the other without music in between.
4.
Click OK or Apply.
Menu Selection Tab
The Menu Selection tab allows for configuration of a voice menu selection that can be made available to callers in queue. When a workgroup queue is controlled by the Advanced Queue Management application, calls in queue will hear a menu prompt. The menu will allow callers to take certain actions based on digit input, and callers can also hear one or more phrases associated with the actions.
Figure 169. Workgroup Configuration, Advanced Queue Management, Menu Selection tab
To set up the Menu Selection:
1.
In the Digits field, select 0 – 9, # or *.
2.
For the highlighted digit, select a Prompt from the phrase list and click Add. You can add one or more prompts, then use the Up or Down buttons to determine the order in which the prompts are played.
3.
Use the list to select one of the following actions, then click OK or Apply.
•
Callback from Queue
•
Transfer to Extension/Other Group
•
Transfer to AA
•
Transfer to Operator
•
Transfer to Outside Number
•
Transfer to Group VM
•
Play prompts
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•
No Action
•
Disconnect
Queue Overflow Tab
The Queue Overflow tab allows for configuration of overflow conditions and actions.
Figure 170. Workgroup Configuration, Advanced Queue Management, Queue Overflow tab
•
Overflow Conditions – Select from any of the following check boxes (if all are checked, the conditions will be followed in order):
–
Calls in Queue exceed – Can be between 0 and 150. This is the number of calls in queue that will cause overflow. For example, 5 calls mean that once a queue has 5 calls in queue, the system will forward the overflow calls according to a specified action.
–
Wait time longer than – Can be between 0 and 200 minutes. This is the time that a call must have been waiting in queue for the call to be overflowed.
–
Service level lower than – Can be between 0 and 100%. This is the percentage of calls in queue longer than service level threshold.
•
Action – Select from one of the following options:
–
Overflow existing call in the queue to (first in, first out)
–
Overflow new incoming calls to (last in, first out)
When either is selected, use the list to select the overflow action:
•
Voice Mail
•
Extension – Select an extension from the list.
•
Workgroup – Select a workgroup from the list.
•
AA – Select the auto attendant to use in the list under the option. AAs are configured in AA
Configuration on the System menu.
•
Operator
•
Outside – Type in the full prefix and phone number, preceded by the trunk or route access code, for example, 915102529712.
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Application Extension Queue Control
When you select Application Extension in the Queue Control panel on the Queue Management tab (and an Application Extension is already configured), use the list to select the desired Application Extension. For
Figure 171. Workgroup Configuration, Queue Management tab, Application Extension Queue Control
Agent Logout Reason Codes
In a workgroup environment, logout reason codes allow agents to specify why they are signing off from the workgroup, and the manager can view that information. If logout reasons are required, the system requests a reason at logout from the phone set and from the Agent application.
The Agent Logout Reason Configuration window lets you require a logout reason, and it provides for defining up to 20 reason codes. A logout history can be tracked and stored for future analysis.
To access this window, select Call Center > Agent Logout Reason Configuration.
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Figure 172. Agent Logout Reason Configuration window
To require logout reasons, select the Logout reason code required check box. If you don’t want to require reason codes, deselect the check box.
To define reason codes, type the associated reason into the text box next to the code you want to associate with the reason.
MaxCall Configuration
The MaxCall Configuration screen is for entering Transmit CID numbers to be used when an agent uses the
MaxCall application to play a phrase to a callee. The campaign names and transmit CIDs you enter here appear in a list on the MaxCall tab in MaxAgent, MaxCommunicator, and MaxOutlook. The agent selects a CID by campaign name before handing a call off to MAXCS. Then MAXCS plays the phrase the agent selected.
Figure 173. MaxCall Configuration screen
•
ID – Campaign IDs are assigned sequentially by the MAXCS system.
•
Campaign Name – The name you give to a calling campaign.
•
Transmitted CID – The caller ID to transmit to the callee when an agent makes a call and uses MaxCall to play a phrase to the callee’s phone.
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Adding a Transmitted CID
1.
Click the Add button. The Campaign and Transmitted Caller ID dialog box opens:
Figure 174. The Caller ID options
2.
Enter a campaign name and a caller ID to transmit when this campaign is chosen by the agent. Click OK.
The campaign names and caller IDs then appear in MaxAgent, MaxCommunicator, and MaxOutlook in the
MaxCall tab list:
Editing a Transmitted CID
1.
Select a campaign and click the Edit button.
Figure 175. Choosing a Caller ID Campaign
2.
Make your changes, and click OK.
Deleting a Transmitted CID
To remove a campaign, select it and click the Delete button. The entry is deleted.
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Callback from Queue
A new feature enables organizations to offer callers the option of receiving a return call instead of waiting on hold in a call queue.
You can customize the specific queue conditions under which a caller is offered the callback option. For example, you can specify that only calls with an anticipated wait time of longer than 25 minutes should be offered the callback option. Or specify that only calls that have at least 9 calls ahead of them should be offered the callback option.
You can also indicate a daily cutoff time for callbacks, so that returned calls can be completed before that workgroup's business day ends.
For return calls, you can specify how many attempted calls to place, how long to wait between those attempted return calls, and a caller ID to transmit with the returned calls.
The Caller’s Perspective
Once you enable and configure the Callback options, callers who reach the threshold for receiving a return call hear various prompts. For example:
"All of our representatives are busy. Rather than remain on hold, you can press 3 to receive a return call.
The wait time for your return call will be approximately <xx> minutes. Press 1 to set up a return call.
Press 3 to remain on hold.
Please say your name. When you have finished speaking, press the # key.
Your name was recorded as <caller name>. Press 1 to continue or press 3 to re-record your name.
You will be called at <incoming phone number>. Press 1 to confirm this number, or press 3 to be called at a different phone number.
Please enter the phone number. When you have finished, press the # key.
You will be called at <phone number> in approximately <xx> minutes. Thank you. Goodbye."
Callers who receive a return call hear various phrases and prompts before they are connected. Following is an example of a successful return call:
"Hello, this is a call back from <company name>. We are returning a call from <caller name>. If we have reached <caller name>, please press 1. To decline this call, press 3.
One moment while we connect you with a representative."
An unsuccessful call back can be the result of a timeout or a declined call. Following is an example of a call that timed out, with no response from the other party:
"Hello, this is a call back from <company name>. We are returning a call from <caller name>. If we have reached <caller name>, please press 1. To decline this call, press 3.
We are sorry that we are unable to reach <caller name>. We will try again later. Goodbye."
Callback Processing Logic
When Callback from Queue is enabled and configured, callers will be offered the option of a return call if the call meets all of the following conditions:
•
The call is in a workgroup queue and it is not a callback call.
•
The workgroup is using Advanced Queue Management and has the Callback feature enabled and configured.
•
The estimated callback time is within the business hours boundary that you have configured for the target workgroup.
•
Either or both of the following conditions must be met:
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•
The call’s expected wait time is longer than the value that you have set for the Expected Wait Time is
longer than xx minutes
option.
•
The call’s position in the queue is greater than the value you have set for the Queue position is greater
than xx
option.
Here is an overview of how Expected Wait-time (EWT) is calculated, to help you understand the Callback feature:
• if total calls (incalls + outcalls) < 10, EWT(expected wait time) = 2.
• if available agent number (not in DND, ready and not in error) = 0, EWT = 2.
• total talk time = incalls talk time + outcalls talk time
•
EWT = ((total talk time * 1.05)/(total calls *available agent number) * in queue position + 59)/60
• if EWT > 240, EWT = 240.
Callback from Queue Licenses
In order to implement the Callback feature, you must have sufficient Callback licenses.
•
AltiGen offers both Callback Seat licenses and Callback Session licenses.
•
If you have purchased both Callback seat licenses and Callback session licenses, when an agent logs in, the seat license will be used if one has been assigned to that user. If no seat license has been assigned to that user, then one session license will be used.
•
In order to log into a workgroup that has been enabled for the Callback feature, the user must have either a seat license or must be using an available session license. If no session licenses are available, then the user will not be able to log into that workgroup and will see an error message.
•
One seat license can apply to multiple workgroups that are enabled for the Callback feature. In other words, if a user belongs to two workgroups that are Callback enabled, the user only needs a single
Callback license. The user does not need a separate Callback license for each workgroup.
Step 1: Assign Licenses and Record Phrases
This feature is a workgroup Advanced Queue Management enhancement.
1.
Acquire the appropriate number of seat and/or session licenses, and assign any seat licenses to the appropriate agents.
2.
If you will not be using the default phrases provided by AltiGen, record custom phrases for the Callback
Reconnect announcement and the Callback phrase. For instructions on recording phrases from your
AltiGen phone or on using professionally recorded phrases, see the section Phrase Management on page
85. Callback phrases are stored in C:\PostOffice\phrases\LangCustom.
Following is a list of some of the phrases that are provided for the Callback feature:
Phrase #
Callback1
Callback
This is a callback for...
You can also press 1 to schedule a callback.
Phrase
Step 2: Configure the Workgroups that will Handle Return Calls
Next, perform these steps on any workgroup that will be handling the return (callback) calls.
Note: The workgroup offering the callback option and the workgroup that will handle the return calls must be on the same server. We recommend that you allocate a single workgroup to handle all callback calls, to simplify reporting of incoming and Callback calls.
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1.
Select Call Center > Workgroup Configuration.
2.
Select a workgroup that will handle the return calls.
3.
On the Call Handling tab, select the option Enable Callback Call Handling. Configure the parameters for the callback calls.
Figure 176. The Callback parameters for the responding workgroup
Parameter
Maximum Attempts
Retry Interval
Maximum Ring Time
Reconnect Announcement
Trunk Access Code
Transmitted CID
Description
The number of times the system will attempt another call if the initial return call fails.
Enter a number between 1 and 10; the default value is 3.
The interval between two return call attempts.
Enter a number between 1 and 10; the default interval is 5 minutes.
The number of seconds that the callback call should ring before drop-ping the call.
The phrase that announces your company name; this phrase is used at the beginning of the return call, to identify the organization that is calling.
The trunk access code that will be used when placing the return call.
The Caller ID that will be transmitted when placing the return call.
This phone number is also logged in the Target Number field of the callback attempt CDR records.
4.
We recommend that you set the Callback Target workgroup to a higher priority if that workgroup’s agents also handle other workgroups at the same time. To do this, on the Call Handling tab, increase the value of the Inter Workgroup Call Distribution parameter.
Figure 177. Inter-Workgroup Call Distribution options
Step 3: Configure the Workgroups that will Offer the Callback Option
Now that you have set up at least one Target workgroup for Callback calls, you can configure any workgroups that will be offering the Callback option.
1.
Select Call Center > Workgroup Configuration.
2.
Switch to the Queue Management tab.
3.
Select the workgroup that will be offering the callback option.
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4.
In the Queue Control group, select Advanced and click Setup. (Note that if your system has many extensions, there may be slight delay while the system opens the Advanced Queue Management panel.)
Figure 178. The Queue Control options
5.
On the Announcement tab, select Queue position, Expected Wait Time, or both.
Figure 179. The Callback option on the Announcement tab
6.
Select the phrase for the Callback Phrase entry. This phrase will be played after each queue update phrase for those calls that meet the callback criteria.
7.
On the Menu Selection tab, add the action "Callback from Queue" to whichever key you've told them to press.
Important:
If you do not specify an action here, the Callback from Queue feature will not be enabled.
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Figure 180. The Callback from Queue action on the Menu Selection tab
8.
On the Callback from Queue tab, specify the following parameters.
Parameter
Enable Callback from Queue when
Target Workgroup
Offer callback option only when the callback time is at least xx minutes before the end of the Target
Workgroup's business hours
Description
To enable callback, you must choose at least one of the two options.
If you check both options, then both conditions must be met for the caller to be offered the callback choice:
•
Queue position greater than xx - Calls that have a queue position higher than the value you specify will be offered the callback option.
Enter a value between 1 and 1,000. The default value is 10.
•
Expected Wait Time longer than xx minutes - Calls with an expected wait time that is greater than the value you specify will be offered the callback option.
Enter a value between 1 and 1,000. The default value is 20 minutes.
Specify which workgroup will be handling the return calls. You can select any workgroup that has the Callback feature enabled.
We recommend that you allocate a single workgroup to handle all callback calls, to simplify reporting of incoming and Callback calls.
If this option is enabled, then the callback option will not be offered to callers if the return call would be within xxx minutes of the Target Workgroup's end of business hours.
For example, suppose that you set this value to 20 minutes. If the return call is calculated to be placed at 4:45pm, and the Target Workgroup's business day ends at 5:00pm, then the caller will not be offered the callback option.
This is because you've specified a minimum 20-minute buffer, but the return call would be placed 15 minutes before the end of the workday.
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Figure 181. The parameters on the Callback from Queue tab
Blocking Area Codes from the Callback Option
You can create a list of area codes that you want to block from return calls. You should configure this for the source workgroup (the workgroup offering the callback option).
1.
In the Call Restriction section of the Workgroup Configuration General tab, check the option Block
Callback Numbers by Area Code, and then click Blocked List. (This option may not be available unless you have enabled that workgroup for callback handling.)
Most North American dial plan long distance area codes are prepopulated. Be sure to check the validity and completeness of this list, as new area codes are added periodically.
Figure 182. The block option for callbacks
2.
Click Add and enter the area code that you wish to block. Click OK. To remove an area code from the blocked list, select it and click Remove. Note that there is a default list of blocked area codes. The list includes the area codes for international calls.
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Figure 183. The Blocked Area Code list
Operational Notes for the Callback Feature
•
Return calls to US Domestic and toll-free numbers are allowed
•
Return calls to International and 900 numbers are not supported
•
Return calls must be directed to a different workgroup than the workgroup that offered the Callback option.
In other words, you cannot direct calls to the same workgroup.
•
In some cases, the announcements may begin playing before the party answers the return call. The announcements will repeat, so that the person hears them.
•
In a scenario such as an internal IT help desk, employees will be calling in, and requesting return calls, from their extensions. If these employees are calling in from other systems, then Enterprise Manager must be set up and employee extensions must be global extensions on the system offering the return call.
•
If the MAXCS server reboots after a return call is scheduled, the return call will not be placed.
•
The workgroup offering the callback option and the workgroup that will handle the return calls must be on the same server.
•
If a return call is requested by an extension on the same MAXCS system and that extension is a virtual extension, then the callback request will be deleted and no call attempts will be made.
•
When a return call is placed and the party accepts the call, the party cannot request a second callback option.
•
If an extension with Multiple Call Waiting enabled is connected to call while waiting on callback, the extension will not receive the return call until the extension is in Idle state.
•
It is the responsibility of the workgroup supervisor/administrator to make sure that the callback workgroup has enough agents available to serve the call at the callback time. If no agents are available, then the return call could potentially go directly to the workgroup's voicemail (depending upon the workgroup's call handling settings). When you enable callback, the Busy Call Handling, No Answer Call Handling, Forward
all calls
, and Group RNA Handling options will be disabled. The Busy Call Handling option is set to Queue.
The other options are set to the Workgroup VM. The Queue management Overflow Forwarding and Quit
Queue
options can only set the target to Workgroup VM. The Queue management page queue control can only be set to Basic.
•
If a callback call is directed to a virtual extension in the same MAXCS system, and the user logs out and back in again, then the callback request will be deleted - no further callback attempts will be made.
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•
Callback data can be found throughout the MAXCS product. Callback data appears in CDRs and in CDR
Search, in various view in MaxAgent and MaxSupervisor, in reports 2206 and 2207 and a new report,
Callback Detail Report
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Using MeetMe Conference
MAXCS provides two different types of conference bridges, Station and MeetMe Conference. Station conferencing is handled from the phone or the desktop client on the fly and requires no configuration in
MaxAdministrator.
The MeetMe Conference is a group conferencing feature that requires the following:
•
A phone meeting needs to be scheduled first by the conferencing host through the client application, or by the system administrator in MaxAdministrator.
•
All participating parties need to dial into the MeetMe conference extension number and enter the Meeting
ID (and, optionally, password) to join the conference.
Hardware requirements:
•
For all OFFICE Chassis series, a 30-port Triton MeetMe conference board (ALTI-CONF-30) is required.
•
Only one Triton MeetMe conference board is supported per system.
•
MAX1000 server has a built-in MeetMe conference (12-port) resource in the embedded DSP. No additional hardware is required.
MeetMe Conference Features
•
Multiple meetings can be held at the same time, as long as the reserved resources do not exceed 30. (For the MAX1000 system, the reserved resources cannot exceed 12, and up to two meetings can be held at the same time. If Gateway Expansion/HMCP are used, all the MeetMe resources should be in the HMCP server, and no MeetMe Conference board should be in a gateway chassis.)
•
Meetings can be set up and administered in MaxCommunicator/MaxAgent and MaxAdministrator.
•
You can set up a single meeting or a meeting that recurs at regular intervals.
•
MeetMe Conference creates an invitation to a meeting, and offers the option to open Microsoft Outlook to send the invitation to people you specify.
•
Option to announce participant's name when joining or leaving the conference. This feature can be configured by the meeting scheduler.
•
Meeting host can Mute/Un-Mute, and drop meeting participants using the desktop client.
•
Meeting host can surrender the meeting control to another extension.
Note: If a conference has configured more than 30 reserve seats, the system will automatically mute all participants to maintain voice quality. The conference host can then un-mute up to 30 participants.
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Setting the MeetMe Conference Extension
Before MeetMe Conference can be used, you must assign a MeetMe Conference extension number. This extension must be dedicated to MeetMe and is the extension that users will always call to join a scheduled meeting. To assign an extension to MeetMe, select PBX > MeetMe Conference Configuration.
1.
Enter an extension number in the MeetMe Conference Extension Number field.
2.
Individual client users can view in the client only the meetings that they have scheduled. The system administrator can view all the meetings that have been scheduled and can manage these meetings. Also, only the system administrator can change the invitation template.
You may want to give someone else the privileges to do these things. In the MeetMe Conference Admin
Number field, you can select the extension of a person to whom you want to give Admin privileges for
MeetMe Conference. That person will see all scheduled meetings in their client application, can manage the meetings, and can modify the invitation template.
3.
Click OK.
MeetMe Conference Window
As system administrator, you may or may not schedule meetings. This may be left to the individual client users.
In the MeetMe Conference window, you can view and manage all the meetings that have been scheduled. You can edit the e-mail template that meeting schedulers may use.
To open the MeetMe Conference window, do one of the following:
•
Click the MeetMe Conference button on the toolbar.
•
Select PBX > MeetMe Conference Management.
This is the same application the clients use. Using this window, you can:
•
Create a one-time or recurring meeting and set its parameters
•
Open Microsoft Outlook to send an e-mail invitation to participate in the meeting
•
Start and stop a meeting
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•
Modify or delete a meeting
•
See meeting ID, subject, scheduler, time, frequency, start time, the last time the meeting started, its status, and the resource being used.
•
Display or hide outdated meetings
•
Modify column display
Working in the MeetMe Conference Window
Click a button to perform a function (for example, create a meeting). The buttons at the bottom of the window are labeled with their function; the buttons at the top of the window perform the same functions. In addition, at the top of the window, you can choose to display outdated meetings by checking the Display Outdated Meet-
ings check box. Deselect the check box to hide outdated meetings.
Select an existing meeting to perform a function on it (for example, to start or stop the meeting). You can select one meeting at a time.
Functions can also be performed on an existing meeting by right-clicking the meeting and selecting from the context menu.
Click a column head to sort by that column. An arrow is displayed that indicates the sort order, ascending or descending. Click again to reverse the sort order.
Use the scroll bar at the bottom of the window to display additional columns, if necessary.
Change column size by clicking and dragging a column border.
Change column order in the current window by dragging a column head to where you want it.
You can open more than one MeetMe Conference window and work with different meetings and displays in each one.
You can double-click a meeting to open the Modify Meeting dialog box.
Using the Calendar Button
The Create Meeting and Modify Meeting dialog boxes use Calendar buttons for date selection. To select a date, click the Calendar button. When the calendar is open, use the Up/Down arrows to select the year, or you can type in a year and then press Enter. Click the Calendar button again to close the calendar.
Creating a Meeting
To create a meeting, click one of the Create buttons.
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The options in the middle panel change, depending on the schedule you select.
The following parameters apply to all meetings:
ID
MeetMe Parameter
Scheduler
Schedule Time
Description
The conference ID is created by the system.
The name of the person scheduling the meeting.
The time the Create Meeting dialog box was opened to create this meeting.
Subject
Reserved Seats
Host
Identifies the subject or type of meeting. What you enter here should be easily identifiable in the meeting list.
Use the Up/Down arrows or type in a number, up to 30, to indicate the number of expected participants. (For the MAX1000 system, expected participants may not exceed 12.)
If a conference has configured more than 30 reserve seats, the system will automatically mute all participants to maintain voice quality. The conference host can then un-mute up to 30 participants.
Select the extension number of the host of this meeting. The host can start and stop the meeting and can mute and drop meeting members.
Frequency
Select the frequency of this meeting from the list. A “weekly” or “monthly” meeting can actually be specified as every 2nd week/month or every 3rd week/ month, and so on.
Middle panel:
Options in the middle panel vary according to the frequency of the meeting. See the sections below this table.
Require Conference
Passcode
Passcode
Announce Participant Name
If you check this, no one can participate who does not enter the conference passcode that you supply.
If you are requiring a passcode, enter it here.
If you want participant names announced when they enter and leave the meeting, check this check box.
Fill in the fields of the Create Meeting dialog box, and click OK. See the following sections for directions on filling in the fields in the middle panel of this dialog box.
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Note: If other scheduled meetings have already reserved resources for the time period, and sufficient resources are not available for the meeting you are attempting to schedule, a message indicates that there is a resource conflict.
One Time Only Meeting
If you select One Time Only from the Frequency list, these are your options in the middle panel:
1.
Specify the duration of the meeting, using the Up/Down arrows (unless this is to be a “permanent” meeting).
2.
If the meeting is to begin as soon as it is scheduled, select Now.
3.
If the meeting is to begin at another time, select On Date, and enter a date and start time. To select a date, click the Calendar button. To select a start time, click the Down arrow and use the slide bar.
Weekly Meeting
If you select Weekly from the Frequency list, these are your options in the middle panel.
1.
In the Duration field, specify the duration of the meeting, using the Up/Down arrows.
2.
In the Start field, specify the start of the meeting by clicking the Down arrow and using the slide bar.
3.
In the Every field, specify how often this meeting is to occur: every week, every other week, every three weeks, and so on.
4.
Check the day of the week on which this meeting will occur.
5.
In the Range of Recurrence panel, use the Calendar button to select a date for the first meeting.
6.
Select End after x occurrences and choose the number of times the meeting is to occur or select End
by and click the Calendar button to specify a date at which the meetings will cease.
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Monthly Meeting
If you select Monthly from the Frequency list, these are your options in the middle panel:
1.
In the Every field, specify how often this meeting is to occur: every month, every other month, every three months, and so on.
2.
In the Duration field, specify the duration of the meeting, using the Up/Down arrows.
3.
In the Start field, specify the start of the meeting by clicking the Down arrow and using the slide bar.
4.
Select either On Date to specify a day of the month by number (for example, the 10th day of the month) or select On to specify a day of the month by name (for example, the first Monday of the month).
If you use On Date, the specified date (for example, the 10th day of the month) may sometimes fall on a weekend day. Check the box Hold during weekend, if the meeting will be held even on a weekend day.
5.
In the Range of Recurrence panel, use the Calendar button to select a date for the first meeting.
6.
Select End after x occurrences and choose the number of times the meeting is to occur or select End
by and click the Calendar button to specify a date by which the meetings will cease.
E-mailing a Meeting Invitation
In the MeetMe Conference window, click the invitation. It will look similar to the following figure.
button to see an automatically generated meeting
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Modifying the E-mail Template
In the Invitation window, you can then click the button to open Microsoft Outlook. The meeting invitation is pasted into a new message in Outlook, and the Outlook Subject field is filled in with “Conference
Call Invitation”. Choose the people to whom you want to send the invitation, make any edits you may want to make, and click Send.
Modifying the E-mail Template
You can edit the e-mail template used by MeetMe Conference to be suitable for your situation. The same template is used for all meeting invitations. To modify it, click the Update Email Template button.
The following variables are included in the template:
•
Schedule
•
Meeting ID
•
Meeting subject
•
Meeting passcode
When you are editing the template, you can choose these variables from the Insert menu to have the specified information automatically inserted where you place it. Users who schedule a meeting can make further edits to the invitation when MeetMe Conference pastes it into Outlook.
Starting and Stopping a Meeting
The meeting host and the MaxAdministrator (Admin) can start and stop a meeting. To start a meeting, select the meeting in the MeetMe Conference window and choose Start. Once the meeting is “started,” the host can log into it (described in the following section).
To stop a meeting before its scheduled duration is over, select the meeting and choose Stop. Manually stopping a meeting frees up resources. Otherwise, the resources will not be freed until the scheduled meeting duration is over.
Continuing a Meeting Beyond Its Duration Time
When the scheduled meeting time is up, the meeting may continue if no other scheduled meeting needs the resources. If another meeting is scheduled and the resources are needed for that meeting, the current meeting is terminated.
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What the MeetMe Conference Host Can Do
In addition to starting and stopping a meeting, the host of a MeetMe conference can:
•
Mute other participants
•
Drop participants from the conference
•
Make someone else the host
Note: If a conference has configured more than 30 reserve seats, the system will automatically mute all participants to maintain voice quality. The conference host can then un-mute up to 30 participants.
To do any of these, either right-click the person’s name in the MeetMe window and choose a command from the menu, or use the buttons at the bottom of the MeetMe tab:
Mute UnMute
Change Host
Drop from
Conference
Note: Changing the host does not affect the Mute setting for the original host or the new host. If the new host was muted before the transfer of hosting duties, that host will have to UnMute himself in order to be heard. Similarly, the original host will still be unmuted if he was unmuted before transferring hosting duties.
Joining a MeetMe Conference
Users calling from an extension can join a meeting by dialing the MeetMe Conference extension number.
Users calling through a trunk must first dial the company number, then the Conference extension number.
Users are prompted to dial the meeting number. If the meeting has not yet been started, the user hears an appropriate message and can try again later.
If a passcode is required, the user is prompted to enter the passcode.
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Location-Based E911 (for Roaming Users)
This feature is designed for roaming users who log into various IP phones at different locations, to use those phones as their extension.
E911 Location ID is supported by Polycom phones.
E911 Caller ID (CID) information is sent to PSAPs (Public Safety Answering Point) when a user calls 911 from an IP phone. A new feature has been added to MAXCS that allows administrators to configure unique E911
Location IDs for specific locations. Each AltiGen IP phone can then be assigned an appropriate E911 Location
ID.
Configuring E911 Location IDs ensures that emergency calls placed from these IP phones:
•
Will call the appropriate PSAP, based upon the phone's physical location.
•
Will transmit the appropriate Caller ID information that will help emergency response teams find the right physical address.
About E911 Location IDs
Each Location ID entry will contain the following information:
•
The phone number of the local PSAP
•
The E911 Caller ID (which is used by PSAPs to determine the correct street address) which is transmitted to emergency personnel when an emergency call is placed
•
A default callback phone number (optional)
E911 Location ID information is associated with the IP phone itself, rather than being tied to a specific user or extension. This way, no matter who logs into an IP phone, the correct E911 Location ID will be transmitted during an emergency call.
Once these E911 Location IDs are configured within MAXCS Administrator, administrators can either push the correct information to the AltiGen IP phones or update the E911 information on each AltiGen IP phone via the phone's menus.
Administrators can also configure whether users must enter a password in order to update an IP phone's assigned E911 Location ID themselves.
Note: When an IP phone is moved to a different location, the Administrator must update the phone's E911
Location ID accordingly, either by pushing it from within MAXCS Administrator or by requesting that a user update the IP phone via the phone's menus.
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Designing E911 Location IDs for Your Organization
Administrators can set up various E911 Location IDs (up to 10,000).
The strategy of how you implement E911 Location IDs will depend upon your unique business.
Some businesses may choose to set up location IDs for each building in a complex, something similar to this:
•
Location ID 10 - Building 315 West
•
Location ID 11 - Building 315 Warehouse
•
Location ID 20 - Building 316
•
Location ID 30 - Building 317
Or, if you have various branch offices, your Location ID schema may look more like this approach:
•
Location ID 1 - San Jose
•
Location ID 2 - Palo Alto
•
Location ID 3 - Redwood City
•
Location ID 4 - Sunnyvale
Another organization may choose to create different E911 Location IDs for each floor of a large office building, similar to this approach:
•
Location ID 1 - Basement/garage
•
Location ID 2 - Floor 1
•
Location ID 3 - Floor 2
•
Location ID 4 - Floor 3
Yet another schema could be to implement E911 Location IDs by the type of use of various building sections, by floor, similar to the following method. This convention uses the first digit as the floor and the second digit as an area within that floor.
•
Location ID 11 - Warehouse
•
Location ID 12 - Floor 1 Lab 1
•
Location ID 13 - Floor 1 Lab 2
•
Location ID 14 - Floor 1 Shipping/Receiving
•
Location ID 15 - Floor 1 Manufacturing
•
Location ID 21 - Floor 2 Test Lab
•
Location ID 22 - Floor 2 Imaging
What's important is to come up with a set of entries that works for you, based upon your type of organization and your geographic specifics.
AltiGen IP Phones Supporting E911 Location IDs
The following AltiGen IP phones are supported:
•
IP 705 (requires firmware version 22A6)
•
IP 710 (requires firmware version 21A6)
•
IP 720 (requires firmware version 23A6)
•
IP 805 (requires firmware version 21, which will appear as “1121” in MaxAdministrator and on the phone itself)
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E911 Hierarchy
If an IP phone does not have the correct firmware version, then you cannot configure an E911 Location ID for the phone until you upgrade its firmware.
New AltiGen IP phones are shipped with the default E911 Location ID entry of 0.
The following phones are not supported:
•
AltiGen IP 600
•
Analog phones connected to the MAXCS server
•
Other 3rd party SIP phones
E911 Hierarchy
The system follows this hierarchy to retrieve E911 information when a user places a 911 call from an IP phone:
1.
The system will transmit the information in the Location-based E911 Caller ID.
2.
If that information has not been assigned, then the system will transmit the extension E911 information that has been configured for the extension in MAXCS Administrator under PBX > Extension
Configuration (see the chapter Extension Configuration on page 179 for details).
3.
If no extension-configured E911 information is available, then the system will transmit the extension’s
Transmitted Caller ID.
4.
If that information is not available, then the system will transmit the extension's DID number (if it is 10 digits or longer).
5.
If there is no DID number assigned to the extension, then the system will transmit the trunk's Transmitted
Caller ID (set in the Trunk Configuration window).
6.
If the area code and phone number have not been configured for the trunk, the System Main Number in the System Configuration window will be sent.
E911 Location ID Configuration Process
Following is an overview of how to configure E911 data for IP phones:
1.
Administrators create a Location ID schema and plan various E911 Location IDs, one for each site, floor, or building.
2.
Administrators configure password protection, so that changing the E911 information for a phone requires
a user to enter a password (optional). See Requiring a Password to Change E911 Location ID on page
3.
Administrators view (and update) the E911 assignments for IP phones. Administrators can either push the updated information to the phones or have users enter the appropriate information on the phone itself.
Creating the E911 Location ID Table
To configure location-based E911 information,
1.
Select PBX > Location Based E911 Configuration.
The panel shows the current E911 entries. To sort the table, click a column heading.
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Figure 184. The Location Based E911 Configuration panel
The following table describes the information shown in this panel.
E911 Location Parameter
E911 LID (Location ID)
E911 CID (Caller ID)
PSAP (Public Safety Answering
Point) Number
Enable Callback
Default callback extension
Description
Description
1-10,000. Enter a unique number for this entry.
The local E911 Caller ID which is used as a Caller ID when the extension places an emergency call.
This field is also unique; each E911 LID must have a different E911
CID.
When this extension makes a local E911 call, this is the PSAP number that will be called.
This setting determines the routing for the call back from the PSAP center.
•
With Callback enabled, the DNIS number of incoming calls is used to match the E911 CID of the 911 call history to determine the callback extension.
•
With Callback disabled, call will be routed to the standard incoming call routing of
MAXCS
, regardless of the E911 call history.
The default setting is Disabled.
The default call back extension; this applies only if Callback is enabled.
•
If the DNIS number from the PSAP does not match the designated
CID from the call history, then the response call will be routed to this callback extension.
•
If you do not enter a default callback extension, the response call goes through the standard incoming call routing.
A brief description for this E911 LID entry.
2.
Below the table, for the Call Back Expiration Time xx Minutes option, set the global expiration time for
E911 response calls. The default interval is 30 minutes.
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Figure 185. The Call Back Expiration field
When Enable Callback is checked, this field sets the expiration time for the routing of response calls.
Response calls will be routed to the last extension that placed an E911 call if the response arrives within that expiration time. After this interval has expired, response calls will be routed using the standard incoming call routing configured in MAXCS.
3.
Click Add.
Figure 186. Add a new E911 Location entry
4.
Note: In order to support E911 LIDs across multiple servers, such that if one server fails the other will correctly handle E911 LID calls, we recommend that each server have an identical list of E911
Location IDs.
Requiring a Password to Change E911 Location ID
Administrators can configure whether users must enter a password in order to update an AltiGen IP phone's assigned E911 Location ID. This requirement can be enabled or disabled for each individual IP phone.
To enable password protection for a phone's E911 information,
1.
Select PBX > AltiGen IP Phone Configuration.
2.
On the General tab, select the phone in the left panel and then check the Enable Protection on E911 checkbox.
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Figure 187. Enable password protection for E911 entry
3.
Once you check that checkbox, the Password field above it becomes active. Enter a password. Your entry will be the password that the user must enter in order to change the Location ID assigned to the phone. Click OK.
Note: You can use the Apply To feature to apply this requirement and the password to multiple phones.
Assigning E911 Locations on AltiGen IP Phones
Administrators can either push the correct E911 Location ID to an AltiGen IP phone, or they can have users enter the correct ID manually on the IP phone itself.
To push an E911 Location ID assignment to an AltiGen IP phone, follow these steps.
1.
Select PBX > Location Based E911 Configuration.
2.
Click View E911 Assignments.
The table lists devices that have, at some point, registered with MAXCS.
3.
Click a column to sort the table. You can also filter the data by choosing All, Active, or Inactive from the
Status Filter
pull-down menu.
Figure 188. The E911 Assignment table
The following table describes the fields in this panel.
E-911 Parameter
E911 LID (Location ID)
Number
Description
The Location ID assigned to this IP phone.
Note: A phone with a Location ID of "0 (Disabled)" either does not have the correct firmware or does not have any E911 Location
ID assigned to it.
The extension number of this IP phone.
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E-911 Parameter
IP Address
MAC Address
Type
Last Login Time
Status
Description
The IP address of this phone.
The MAC Address of this IP phone. Polycom phones will not display a
MAC address.
The type of phone.
The last time that this IP phone logged into the
MAXCS
system.
The current state of this IP phone. You can filter the data in this list as needed.
•
Active - The IP phone is currently registered with the
MAXCS system.
•
Inactive - The IP phone is not currently registered with the
MAXCS system (the phone may have been unplugged or was logged out of the system).
4.
To change the E911 Location ID for an IP phone, select the entry and click Change LID.
Note: If you cannot click the Change LID button for that phone (in other words, if the button is disabled), this indicates that the phone does not have the correct firmware; you must upgrade the firmware on the AltiGen IP phone before you can set the phone's E911 Location ID..
5.
Make your changes and click OK. The data will be pushed to the phone, and the phone will automatically re-register to apply the updated configuration.
6.
To remove an E911 LID assignment for an extension, select the entry and click Delete. (Note that you can delete assignments only for inactive AltiGen IP phones.)
To have a user update the E911 Location ID on an IP phone, provide the user with the appropriate Location
ID number. Instruct the user to open the phone's menu, choose System > E911 LID, and enter the Location
ID number.
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Network Configuration Guidelines for VoIP
Real-time applications such as voice communications require a networking environment that meets certain requirements to deliver and maintain good voice quality. Follow these network configuration guidelines when using MAX Communication Server’s VoIP features.
ISP/Intranet Quality of Service (QoS)
•
If you subscribe to the public IP network or use your own Intranet, make sure the maximum network delay is less than 100 milliseconds.
•
Also, the typical packet loss rate should be less than 1 percent.
Virtual LANs
MAXCS supports virtual LANs in accordance with IEEE 802.1Q. A virtual LAN (VLAN) segments an Ethernetbased network into different logical networks that provide different services such as data service and voice service. It also defines broadcast domains to reduce network traffic load. It provides a managed network environment to run voice and data together smoothly.
The IEEE 802.1Q header includes IEEE 802.1p, a standard method for assigning priority to packets traversing a network. It works with the Ethernet MAC (Media Access Control) header at the data link layer. The managed switches in a network are responsible for differentiating packets based on their priorities and processing them in different orders.
Requirements:
•
MAXCS or above with two NICs for 802.1Q VLAN
•
MAXCS or above for 802.1p
•
NIC support 802.1p for 802.1p
•
The following AltiGen IP phone firmware:
–
VLAN: 2xA8 and boot code version 12 or above
–
802.1p: 2xA8 (MAXCS firmware)
•
Layer 2 managed switch
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Ethernet II Framing Header
The Ethernet II framing header is defined as follows, with 802.1Q VLAN tag and 802.1p priority bits:
For 802.1Q VLAN-tagged Ethernet frame, the Tag Protocol Identifier (TPID) or Ethernet Type is set to 0x8100.
The next 16 bits defines the VLAN and QoS bits:
•
Priority Code Point (PCP): a 3-bit field which refers to the IEEE 802.1p priority. It indicates the frame priority level from 0 (lowest) to 7 (highest), which can be used to prioritize different classes of traffic (voice, video, data, and so on).
•
Canonical Format Indicator (CFI): a 1-bit field. If the value of this field is 1, the MAC address is in noncanonical format. If the value is 0, the MAC address is in canonical format. It is always set to zero for
Ethernet switches.
•
VLAN Identifier (VID): a 12-bit field specifying the VLAN to which the frame belongs. A value of 0 means that the frame doesn't belong to any VLAN; in this case the 802.1Q tag specifies only a priority and is referred to as a priority tag. A value of hex FFF is reserved for implementation use. All other values may be used as VLAN identifiers, allowing up to 4094 VLANs.
Only port-based VLAN is supported in MAXCS, which means the VLAN is assigned in the switch port and managed in the switch internally. The end device, like the MAXCS NIC and IP phone ARM MAC port, does not need to tag the packet with VLAN so there is no software implementation on the end device. MAXCS can use two NICs and connects them to the switch ports with a different VLAN assigned so the network traffic can be separated. However inside the IP phone, the firmware programs the Ethernet switch to assign and manage the different ports with different VLAN IDs. The IP phone user can configure the IP phone port with voice VLAN
ID and PC port with data VLAN ID. Different VLANs use a different IP network. Below is a typical VLAN setup:
The NIC in both the MAXCS server and the IPTalk client (used with MaxCommunicator/MaxAgent) must support 802.1p. To see if the NIC supports the 802.1p feature, open the NIC’s Properties dialog box and select the Advanced tab. See if the "QoS Packet Tagging" property is in the Property list. (Different NICs have different properties and may display a different property name for the 802.1p feature.) If the NIC supports the
802.1p feature, the default value is Disabled and you can change this value to enable 802.1p as seen in the following figure.
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Virtual LANs
For 802.1p, eight different classes of service are available, expressed through the 3-bit user priority field in an
IEEE 802.1Q header added to the frame. The IP phone tags the voice packet from the ARM processor with a user configured value.
Specifying the Priority Value
The server side configuration is located in the HMCP board's Board Configuration settings or the VoIP board’s Board Configuration > Advanced setting.
In MaxCommunicator/MaxAgent, the configuration is in the IPTalk configuration screen:
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Enabling VLAN
VLAN can be enabled and configured in the AltiGen IP phone: Network > Enable VLAN > Yes.
After enabling, set VID:Phone and VID:PCPort IDs.
It can also be configured in MaxAdministrator in the AltiGen IP Phone Configuration screen, General tab.
WAN Bandwidth
and frame sizes.
•
The Jitter Buffer should be adjusted according to the bandwidth allocated to data traffic. For example, a long Ethernet packet (approximately 1500 bytes) traversing through a WAN which is allocated with 256 kbps of data traffic bandwidth will take about 50 milliseconds. The Jitter Buffer value should be set to this
WAN link transmission delay plus the typical network jitter delay. To configure the Jitter Buffer, in Enterprise Manager (VoIP > Enterprise Network Management) click the Codec button.
•
If you have heavier data applications running concurrently, the bandwidth reserved for data traffic should be increased.
•
If your router supports multilink or TCP fragmentation, configure your WAN router to user smaller packet sizes, for example, 500 bytes.
WAN Router Configuration
•
The router that connects your LAN and the WAN should support priority queuing.
•
Configure the router so that the IP/UDP packets being sent to and from an IP station have higher priority than the packets generated by other stations on the same network. Consult your router manufacturer for more information on setting up this configuration.
Firewall Configuration
If a firewall is used to protect your network access security, reconfigure the firewall to open up TCP and UDP
voice to pass through the firewall freely.
Network Using NAT
You are probably using NAT if both of the following conditions apply:
•
Your MAXCS ACM server's IP address matches any of the following numbers (where x is any number from 0-255):
•
10.x.x.x
•
172.16.x.x to 172.32.x.x
•
192.168.x.x
•
You are able to connect to the Internet directly without using a proxy server.
•
Contact your router/firewall vendor to obtain a software update for your networking equipment, or obtain routable address space from your Internet provider. If you are unsure whether or not you are using NAT, contact your router/firewall vendor or Internet provider.
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Network Configuration Guidelines for AltiGen IP Phones
Network Configuration Guidelines for AltiGen IP Phones
The following guidelines (specific to AltiGen IP phones) should be taken into consideration before you configure your network for use with NAT.
•
DHCP is recommended to reduce the risks for duplicating IP addresses. MAXCS provides seamless support for AltiGen IP phones using dynamic IP addresses. Select Dynamic IP address for IP Extensions in the Max Admin’s Extension Configuration window.
•
A switch is required; VoIP quality can be adversely affected if a hub is used.
We recommend the following settings on the MAXCS server:
•
Large Send Offload VZ (IPv4) – Disable
•
Receive Side Scaling – Disable
•
IP Security (IPSec) Offload – Disable
•
TCP Checksum Offload IPv4 – Disable
•
UDP Checksum Offload IPv4 – Disable
Configuration Guidelines for NAT
Note: This section applies only to AltiGen IP phones or IPTalk integrated with MaxCommunicator or
MaxAgent.
The section discusses the configuration guidelines when MAXCS is behind NAT (Network Address
Translation) and communication to AltiGen IP phones, IPTalk, or another MAXCS is over WAN. AltiGen SIP phones support NAT traversal, which does not require special settings on the NAT router at the remote site.
Due to SIP protocol, which puts the IP address information in the TCP/IP payload, the NAT router requires some SIP protocol implementation to correctly handle the SIP traffic and translate the private IP address into a public IP address. Not all NAT routers have this kind of implementation, or have it implemented correctly (SIP
ALG). If the NAT router does not support SIP ALG properly, you need to check Enable SIP NAT Support and
Enable SIP NAT Support in Enterprise Manager on the IP Networks tab. Also, SIP ALG must be disabled in the NAT router to avoid the conflict between SIP ALG and SIP NAT support provided by MAXCS..
The following sections illustrate a private network configuration and a VPN configuration.
For information on setting up VoIP traffic forwarding for NAT and configuring MAXCS behind NAT, see
“Configuring MAXCS Behind NAT” on page 337.
Private Network Configuration Example
(MAXCS with private IP address and behind NAT)
Only the private IP address is used in a private network. The public router will not route the packet that has a private IP address as its destination. (All IP addresses beginning with 192.168.x.x, 10.x.x.x, or 172.16.x.x to
172.32.x.x are private IP addresses.)
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Figure 189. MAXCS behind NAT
This figure shows a private network, 192.168.1.0, where MAXCS is installed and running on a host with a private IP address 192.168.1.2.
Router 1 is a NAT router. The local IP phones – IP Phone 1 and IP Phone 2 – use the private IP addresses
192.168.1.100 and 192.168.1.101, respectively. There are two remote IP phones: IP Phone 3 with a private IP address 192.168.2.100 connects to the Internet via Router 2. Router 2 can also sit behind a DSL/Cable Modem.
Example Setup: Corporate LAN
•
MAXCS – MAXCS is installed (private IP address 192.168.1.2). The public IP address of Router 2 should be configured as the IP address of this IP extension in MAXCS. If it is changed dynamically, then assign a dynamic IP address configuration for that extension.
•
Router 1 – Router 1 is a NAT router. You need to set up the SIP port forwarding for this NAT router from
169.254.56.169 to the private IP address of MAXCS 192.168.1.2.
Example Setup: Remote IP Phone Using NAT
•
IP Phone 3 – When configuring remote IP Phone 3, you should set up the MAXCS IP address to Router
1’s public IP address — 169.254.56.169.
•
Router 2 – No special configuration is needed for Router 2. Also, more than one AltiGen SIP phone can sit behind Router 2.
Example Setup: SIP IP Call from Another MAXCS on the Internet
Another MAXCS can make an SIP IP call to this MAXCS by calling the MAXCS's public IP address, which is
169.254.101.2.
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VPN Network Configuration Example
(Connecting to MAXCS with VPN)
Configuration Guidelines for NAT
Figure 190. MAXCS with VPN
In a multi-site configuration, VPN can be used to provide a secured tunnel between the remote sites and the corporate site.
LAN and branch office LAN. The VPN tunnel connects the two private networks such that the two networks access each other with a private IP address.
In the corporate network, MAXCS is installed on a host with private IP address 192.168.1.2.
Both Router 1 and Router 2 are VPN-capable and compatible with each other. (It is recommended that the routers come from same vendor.) A VPN tunnel exists between these two routers. The local IP phones – IP
Phone 1 and IP Phone 2 – directly connect to the corporate network with private IP address 192.168.1.100 and
192.168.1.101. And the three remote IP phones – IP Phone 3, IP Phone 4 and IP Phone 5 – connect to the branch office network with private IP addresses 192.168.2.100, 192.168.2.101 and 192.168.2.102, respectively.
Example Setup: Branch office LAN
•
IP Phone 3, IP Phone 4 and IP Phone 5
When configuring the remote IP phones – IP Phone 3, IP Phone 4, and IP Phone 5 – you should set up the AW address to use the MAXCS IP address.
For the VPN Tunnel between the Two Private Networks:
You must set up a VPN tunnel to connect the two private networks. The VPN setup procedure may be complicated and is generally performed by a professional IT technician.
The following minimum guidelines need to be considered for setting up the VPN tunnel:
•
WAN Bandwidth – Should be greater than the aggregate of maximum VoIP session bandwidth usage.
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•
QoS – If the IP WAN network provides QoS (Quality of Service), it should be configured to honor VoIP
RTP packet transmission.
An easy example for a VPN resolution is with the Linksys EtherFast VPN router
1
. Router 1 and Router 2 are routers supporting VPN. When configuring these VPN routers, the following information is needed.
(Also, please refer to the Router's User Guide for more detailed information.)
Router 1's Settings
Local Secure Group:
(specifies the local network which can access the VPN tunnel at the corporate network)
Remote Secure Group:
(specifies the remote network which can access the VPN tunnel at the branch office network)
Remote Security Gateway:
(specifies the public IP address of the remote gateway which can access the VPN tunnel at the branch office)
Subnet IP:
192.168.1.0
(Corporate Network)
Subnet IP:
192.168.2.0
(Branch Office Network)
63.224.32.34
(Router 2’s public IP
Address
Router 2's Setting
Router 2's public IP address should be a fixed IP address.
Local Secure Group:
(specifies the local private network in the branch office, which can access the corporate network through VPN)
Remote Secure Group:
(specifies the corporate network, which can be accessed by stations in this local private network through the VPN tunnel)
Remote Security Gateway:
(specifies the public IP address of the corporate VPN-enabled gateway)
Subnet IP:
192.168.2.0
(Branch Office Network)
Subnet IP:
192.168.1.0
(Corporate Network)
169.254.56.159
(Router 1’s public IP
Address
Subnet Mask:
255.255.255.0
Subnet Mask:
255.255.255.0
Subnet Mask:
255.255.255.0
Subnet Mask:
255.255.255.0
1
Linksys is for reference only. AltiGen has not certified this product or any other router at this time.
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Enterprise VoIP Network Management
The VoIP-related aspects of both single-server systems and multi-site VoIP domains are configured in
Enterprise Manager, available from the VoIP menu or the Windows Start menu.
In addition, multi-site VoIP domain management – including directory synchronization and routing – is handled here.
Notes
•
With MAXCS Private Cloud systems, Enterprise Manager’s domain-related features require a VPN or
MPLS to the cloud service.
•
A multi-site installation requires an Enterprise License.
•
A security enhancement has been added for the AltiGen Java Service Loader.
Prior to Release 8.0, you could use AltiGen Java Service Manager to connect a remote machine's Java
Service Loader with the default password, if the remote server's administrator had not changed the password.
In Release 8.0, however, the new Java Service Manager can connect only to the local system. Also, earlier versions of the MAXCS AltiGen Java Service Manager will not be able to connect to the MAXCS 8.0
AltiGen Java Service loader.
This change will prevent you from running Enterprise Manager from a LAN or WAN connection.
The only way to access Enterprise Manager now will be from the server itself, via Remote Desktop.
For a single-system installation, only the following VoIP configuration elements in Enterprise Manager are relevant and are discussed in the first part of the chapter:
•
Codec Profile – Create codec profiles that use different settings for jitter buffer size and packet length.
Codec profiles can be assigned to different types of VoIP connections, as defined in the IP dialing table and IP codec assignment table.
•
VoIP Bandwidth Use – Define the maximum VoIP sessions using different codecs on a public Internet or a private intranet data pipe.
•
NAT Support – Configure VoIP NAT traversal when the server is behind NAT using a private IP address.
•
IP Dialing Table – Define IP dialing digits and codec for VoIP dialing to other AltiGen systems or certified third-party IP devices.
•
IP Codec Table – Define the codec and data pipe for AltiGen IP phones and SIP trunking service.
Note: AltiGen IP phones 705, 710, and 720 do not support G.722 or G.711 A-Law.
For a multi-site installation, you can manage the above configurations for all your VoIP domain servers from
Enterprise Manager.
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Understanding VoIP Bandwidth Requirements
Along with the above configurations, the multi-site administrator will use Enterprise Manager and the VoIP menu in MaxAdministrator to do the following:
•
Create the VoIP domain
•
Define the VoIP domain Master
•
Join servers to the VoIP domain
•
Manage VoIP domain users
•
Define global least cost routing
Understanding VoIP Bandwidth Requirements
Before starting VoIP related configurations, it is helpful to have some understanding of VoIP bandwidth require-
The data network bandwidth required to carry VoIP depends on the following factors:
•
Codec and Compression – This is the encoding of analog voice to digital form, decoding of digital form to analog wave form, and compression of digital form to a smaller size. MAXCS supports several type of codec: G.711, G.722, G.729, G.723.1.
•
Packet Length (Frame Size) – The size of the voice frame data (payload) transmitted in a packet. For
G.711, G.722, and G.729, you have choice of 10, 20, and 30 ms lengths. For G.723.1, the packet length is a fixed 30 ms. A larger packet length decreases the transmission overhead. However, it will increase the latency and have a negative effect on the voice quality if a packet is lost during transmission. For
G.711, G.722, and G.729, 20 ms is efficient and recommended.
•
IP Header – The IP/UDP/RTP header adds 40 octets per packet. With a packet length of 20 ms, the IP headers will require 16 kbps of bandwidth in addition to whatever codec is being used.
•
Transmission Medium – In order to travel through the IP network, the IP packet is wrapped in another layer by the physical transmission medium. The transmission medium, such as Ethernet, will add its own header, checksums, and spacers to the packet. With a packet length of 20 ms, the transmission medium requires additional 15.2 kbps of bandwidth to carry the packets to their destination.
•
Silence Suppression – You can suppress the transmission of data during periods of silence. This can reduce the demand for bandwidth by as much as 50 percent. However, it may have a negative impact on the voice quality. Some users may feel the conversation is not "natural" when artificial comfort noise is generated during periods of silence.
The following table lists bandwidth requirements for various transmission media with different codecs and frame sizes. It assumes silence suppression is not turned on.
Codec
G.711/G.722
64
G.711/G.722
64
G.711/G.722
64
G.729
8
G.729
G.729
G.723.1
8
8
6.4
Voice
Encoding
(kbps)
Frame
Size
10 ms
20 ms
30 ms
10 ms
20 ms
30 ms
30 ms
PPP
(kbps)
100.8
82.4
76.3
44.8
26.4
20.3
18.7
Frame
Relay
(kbps)
102.4
83.2
76.8
46.4
27.2
20.8
19.2
Ethernet
(kbps)
126.4
95.2
84.8
70.4
39.2
28.8
27.2
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Opening Enterprise Manager
VoIP Bandwidth requirement for WAN connection varies depending on the type of WAN. Bandwidth requirement typically is less than Ethernet requirement.
Opening Enterprise Manager
To open Enterprise Manager, use one of the following methods:
•
For a single-system installation without a Domain Master, this first method is recommended. From Max-
Administrator, select VoIP > Enterprise Network Management. Enterprise Manager opens without a login dialog box.
•
For multisite VoIP domain management, from the Windows Start menu, select All Programs > MAXCS
> Enterprise Manager. A login screen opens. (With this method you can log in to the Domain Master from any member system or remote desktop.)
Figure 191. The Login panel for Enterprise Manager
User name Password
DomainAdmin
(Logging in as DomainAdmin gives you rights to change the entire Enterprise
Manager configuration.)
Default: 22222. You can change the password in Enterprise
Manager.
Note: This password is not the same as the MaxAdministrator password.
Admin@domain master IP address
(A Site Admin who logs into the Domain
Master in this way has the same rights as DomainAdmin.)
Admin@member server IP address
(A Site Admin who logs in this way can make changes on this member server only.)
Enter MaxAdministrator password
Enter the MaxAdministrator password for the member server
Login Domain Via
Server
Enter the domain master’s IP address
Enter the domain master’s IP address
Enter the member server’s IP address
Warning! If your MAXCS system is using dynamic IP addressing, you will see the following warning message when launching Enterprise Manager. Please check the Internet Protocol (TCP/IP) Properties of your server NIC interface and assign a fixed IP address to this server.
Figure 192. Example of a log-in error message
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Overview of Enterprise Manager
When multiple systems are added to the VoIP domain, all member systems need to have both Route Access
Code and IP Trunk Access Code configured. If one or more member systems are not configured properly, a message opens.
Multisite routing may fail if Route Access Code and IP Trunk Access Code are not configured.
Overview of Enterprise Manager
After you log in, the Enterprise Manager window opens.
Change Password
Configure Codec Profiles
Display Servers
Configure Users, Departments Configure Global Least Cost Routing
Figure 193. The Enterprise Manager window
Click a tab to view or configure settings on that tab. Information on a tab is related to the selected server. Click buttons in the toolbar to perform configuration tasks. Click a column heading to sort by that column.
Configuration Buttons
•
Servers – Displays the VoIP domain name, servers in the system, and server ID length. Lets you add/ remove servers and change the VoIP domain master. Lets you re-route outgoing calls of global extensions and redirect IP phones. Displays the configuration and informational tabs listed in the next section.
•
Codec – button lets you configure individual codec profiles – silence suppression, codec, DTMF delivery, enable/disable SIP early media, SIP transport, jitter buffer range, and RTP packet length.
•
User – Displays information about extensions in the VoIP domain and lets you change an extension to global or local and relocate an extension.
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Setting VoIP Codec Profiles
•
Department – Lets you define departments in the VoIP domain and assign extensions to departments.
•
Global LCR – Lets you add E.164 number patterns and specify source and target sites.
Tabs Displayed with the Servers Button
•
Information tab displays information about the selected site and lets you configure a PSTN number for global extension rerouting as a failover when the TCP/IP network is down. You may also assign an alternate server to which to redirect global AltiGen IP phones when their primary server is down.
•
IP Networks tab defines IP networks and the bandwidth information for an MAXCS site. Bandwidth usage control for Internet and intranet can be set up here. If the bandwidth usage exceeds the maximum setting, the call will not be established.
•
IP Dialing Table tab defines the IP dialing table for a MAXCS site. Specified information here includes a codec profile and a protocol (SIP) for the communication from this site to the selected site.
•
IP Codec tab lets you specify an inter-gateway codec and define IP device ranges to which you can assign a codec profile.
•
Number Plan tab displays the number plan information that is set up in MaxAdministrator, System
Configuration window, Number Plan tab.
Changing the Enterprise Manager Password
Only a person with DomainAdmin rights can change the Enterprise Manager password. To change the password, click the Password button at the top of the Enterprise Manager window.
Figure 194. The Change Password panel
Enter the old password, and the new password. Confirm the password, and click OK.
Setting VoIP Codec Profiles
The codec setting is profile-based. For different IP addresses and protocols, a different preferred codec can be used. Each codec profile can have its own codec (G.711, G.722 on Softswitch or Cloud, G.723, G.729), packet length, and jitter buffer. The codec profile can be assigned to connectivity with a remote server, IP phone or other VoIP device.
By default, the following IP address ranges (private IP addresses) will use G.711 codec:
•
192.168.0.0 to 192.168.255.255
•
172.16.0.0 to 172.31.255.255
•
10.0.0.0 to 10.255.255.255
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Setting VoIP Codec Profiles
Figure 195. The IP Codec sub-tab showing devices and their assigned codecs
To open a window where you can set or modify codec profiles, click the Codec button in the Enterprise
Manager toolbar.
Figure 196. Codec profile setting window in Enterprise Manager
Named codec profiles are listed on the left.
To create a new profile,
1.
Click the Add button.
2.
Name the new profile and click OK. Make your changes, and then click OK.
The next two tables describe the Codec parameters.
Codec Parameter
Codec Profile Table
Description
Lists codec profiles by name. Select a profile in the table to modify its settings, then click Apply in the panel where you made the changes.
Click the Add button to add a codec profile. Click the Remove button to remove the selected profile. You cannot remove the Default profile.
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Setting VoIP Codec Profiles
Codec Parameter
Name
Codec
G.711 / G.723 / G.729
Silence Suppression
Video Codec
DTMF Delivery
(Applies to SIP protocol only)
SIP Early Media
(Applies to SIP protocol and SIP trunk only)
Description
Name of the codec profile. You can modify the name, and click Apply.
The Default profile name cannot be changed.
There are several options:
• G.711 Mu-Law
• G.723.1
• G.729
• G.722 (supported on Softswitch and Cloud deployments only)
G.711 provides toll quality digital voice encoding, and G.723 and
G.729 use low rate audio encoding to provide near toll quality performance under clean channel conditions.
G.722 is a wideband audio codec. It provides higher voice quality than more narrow-bandwidth codecs such as G.711.
Use the Add and Remove buttons to move codecs between the
Available Codec
and the Selected Codec lists. Codecs you place into the Selected Codec list will appear in the SDP Codec Capability list for codec negotiation.
Prioritize the selected codecs by using the Up and Down buttons.
Note: While Polycom phones support G.722, AltiGen IP phones 705,
710, and 720 do not support G.722 or G.711 A-Law. In mixedphone environments, make sure you include G.711 Mu-Law,
G.723-1,or G.729 in the Selected Codecs list.
When silence suppression is enabled, and silence is detected during a call, MAXCS stops sending packets to the other side. This decreases the bandwidth requirement, however the voice quality may be degraded slightly. These are system-wide settings.
Check the Enable H.264 Codec option to enable video support for
P2P Polycom VVX phones.
Note: MAXCS does not support video over SIP Tie trunks or SIP trunks.
This option is only for Softswitch deployments.
SIP INFO is used to deliver DTMF.
RFC 2833 – The DTMF pay load is embedded with RTP. Most 3rdparty SIP gateways support this standard.
Note: The codec profile assigned to the SIP-TIE Trunk must use
RFC 2833 (DTMF) in order to support Polycom phones. We recommend that you configure a single codec for SIP Trunk interfaces.
In band – If DTMF tone is delivered over the voice band. It’s not reliable over G.711 codec and will not work over G.729/G.723 codec
SIP Early Media allows two SIP devices to communicate before a SIP call is actually established. It is important for interoperability with the
SIP trunk carrier’s PSTN gateway. If SIP Early Media is not checked, the caller may not hear the exact ringback tone provided by the CO
(the caller may not hear any ringback tone at all).
When using a SIP trunk as a MaxMobile trunk, you must enable the
SIP Early Media option for the SIP trunk.
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Setting VoIP Codec Profiles
Codec Parameter
SIP Transport
Description
There are several SIP Transport options. Note that security options can be configured for individual IP phone extensions in the IP Phone
Configuration screen. (For more information on security settings, see
“SIP Transport” on page 221.) Extension-level configuration takes
precedence over a codec profile that is assigned in Enterprise
Manager. See the next section.
UDP – User Datagram Protocol is a communications protocol that offers a limited amount of service when messages are exchanged between computers in a network that uses the Internet Protocol (IP).
Note: All SIP trunks must use UDP.
TCP – Transmission Control Protocol is a set of rules (protocol) used along with the Internet Protocol (IP) to send data in the form of message units between computers over the Internet. TCP is known as a connection-oriented protocol, which means that a connection is established and maintained until such time as the message or messages to be exchanged by the application programs at each end have been exchanged. TCP is responsible for ensuring that a message is divided into the packets that IP manages and for reassembling the packets back into the complete message at the other end.
Note: AltiGen phones do not use TCP.
TLS/SRTP – Adds Secure RTP to Transport Layer Security to secure
SIP-associated media.
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Click Advanced to access additional advanced parameters.
Setting Codec Priority
Figure 197. The Advanced Codec Profile Settings panel
Advanced Codec
Parameter
Description
G.711 / G.722 / G.723 /
G.729 Jitter Buffer Range
(ms)
G.711 / G.722 / G.729
RTP Packet Length (ms)
Indicates the delay, in milliseconds, used to buffer G.711, G.722,
G.723, and G.729 voice packets received from the IP network. Voice packets sent over the IP network may incur different delays due to network load or congestion. The jitter buffer helps to smooth out the delay variation in the arriving voice packets and maintain voice quality at the receiving end.
The default values for G.711 are 10 min. to 100 max milliseconds.
The default values for G.722 are 10 min. to 100 max milliseconds.
The default values for G.723 are 30 min. to 480 max milliseconds.
The default values for G.729 are 10 min. to 480 max milliseconds.
Lets you configure the length of the RTP packets for G.711, G.722, and G.729 in milliseconds. The RTP packet length can be set to 10,
20 or 30 milliseconds.
The smaller the packet length, the larger the bandwidth required.
Setting Codec Priority
You can set codec priorities for each Codec Profile.
During codec negotiation, the highest-priority codec that the receiving end supports will be used. Therefore, you should sort the Selected Codec list by priority, from top to bottom. Use the Up and Down buttons to do this.
Prioritize this profile’s codecs by arranging codecs top (highest priority) to bottom (lowest priority)
Figure 198. Codecs in the Selected Codec and Available Codec lists
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About the G.722 Codec
Codec Negotiation
The logic for negotiating codec capabilities varies for various types of connection.
Codec Negotiation - Server to Client
With a server-to-client connection, the highest priority codec in the server's codec list that is supported by the client will be assigned.
Following is an example of a typical setting for SIP devices on a LAN: G.722, then G.711 Mu-Law. This hierarchy allows Polycom phones and the AltiGen IP-805 phone to connect via G.722. AltiGen IP phones will connect with G.711.
Figure 199. Codec negotiation example: SIP devices on a LAN
The next example shows a typical setting for SIP devices on a WAN: First G.729, the G.723.1, and then G.711 u-Law. This hierarchy allows remote SIP devices to use the compressed codec G.729. Remote IPTalk instances use G.723.1, and G.711 is used for other remote devices that do not support the other two codecs.
Figure 200. Codec negotiation example: SIP devices on a WAN
Codec Negotiation - Server to Server
With server-to-server connections, the responding device determines the codec from the common codecs that both of the devices accept. Usually, the responding device selects the first preferred codec from among the common list. We recommend that you assign a preferred codec on both servers, so that both incoming and outgoing events use the same codec.
If you are not sure of the codecs supported by your SIP Trunk provider, then we recommend that you configure a single codec (your preferred codec) for SIP Trunks.
About the G.722 Codec
MAXCS supports the G.722 codec on Softswitch and on Cloud deployments. G.722 is a wideband audio codec. The G.722 codec provides higher voice quality than more narrow-bandwidth codecs such as G.711.
MAXCS supports G.722 64 kbps only.
The CPU processing requirement for G.722 is similar to G.723 and G.729. It is designed to use the same combo codec resources as G.723 and G.729.
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Assigning Codec Profiles to IP Addresses
The G.722 codec is supported in HMCP media servers only. AltiGen's telephony boards, including the Triton and Vision boards and the Max1000 and Max2000 boards, do not support the G.722 codec.
G.722 is supported by the following devices:
•
AltiGen IP 805 phone - G.722 64 kbps ONLY
•
Polycom phones
•
SoundPoint IP-335, IP-550, IP-560, IP-650, IP-670
•
SoundStation IP-6000, IP-7000
•
VVX 300, VVX 310, VVX 400, VVX 410, VVX500, VVX 600
•
SIP tie trunks
The G.722 codec is part of combo codec and is controlled by license.
Note: When adding additional combo licenses, the system will also increase the RTP ports it uses and will use these new ports. If these additional ports are not added to the firewall, then calls will not have audio.
The number of total codecs is shown in the License Information panel.
Figure 201. The License Information panel
Assigning Codec Profiles to IP Addresses
You can specify what codec profile to use when connecting to the following VoIP devices:
•
IP phones on the LAN
• a remote IP phone over WAN
• a remote AltiGen system over WAN
•
SIP Trunk service provider over WAN
• multiple gateways on the LAN
The codec profile assigned in the IP Device Range table (shown below) supersedes the codec profile defined in the IP dialing table if the IP address is duplicated in both tables.
The SIP transport assigned to an extension in the IP Phone Configuration screen takes precedence over a codec profile with a different SIP transport assigned in Enterprise Manager. If the IP extension supports TLS and the codec profile does not, then the IP extension policy holds. That way you can configure a range of IP addresses in the IP dialing table or IP codec, and have only a few IP addresses/extensions support TLS.
If the IP extension has not configured TLS as its transport, and the codec profile supports it, then the codec profile policy holds.
To set IP address ranges and assign codec profiles to them, in Enterprise Manager, on the Servers tab, click the IP Codec sub-tab.
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Managing IP Networks
Figure 202. The IP Codec sub-tab
By default, all private addresses are set to G.711 codec only. You can add individual IP addresses and address ranges and assign a codec to each.
Add IP Addresses and Address Ranges and Assign a Codec
1.
Click the Add button in the IP Device Range panel (the panel on the right).
2.
Enter an IP address range (for dynamic IP addressing), or enter the same address in each field if this is a static address. You cannot use the minimum and maximum values (0.0.0.0. and 255.255.255.255).
3.
Click OK.
If you have multiple gateways controlled by a MAXCS host system, you need to configure an Inter Gateway
Codec profile.
Set the Codec for a Connection among Gateways in the Same MAXCS Server
1.
Select a server in the Global Server Location list on the left side of the window.
2.
In the Codec field, select the codec to use for a connection to this server from the list.
Managing IP Networks
In MAXCS, Instead of specifying the maximum number of sessions per each codec, as in previous releases, you specify the maximum number of sessions per bandwidth range.
The actual network bandwidth required for the bit rate range is used for the bandwidth calculation. 33 to 64
Kbps bit rate is translated into 100 Kbps network bandwidth, which includes LAN and IP/UDP/RTP header overhead. 16 to 32 Kbps and under 16 Kbps are translated into 65 Kbps and 40 Kbps, respectively.
The maximum number of allowed G.711 and G.722 codecs appears by default. You can lower these numbers by clicking the down arrow until you reach the desired amount.
After the maximum number of sessions has been reached, the call negotiated with this codec bit rate will be rejected. Also, after the total bandwidth utilization of all the codecs exceeds the value specified in Bandwidth for VOIP parameter, new calls cannot be established.
You configure this on the IP Networks sub-tab in Enterprise Manager.
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The tab allows you to specify the following limits:
•
Bandwidth for VoIP sessions in the Public Pipe
•
Bandwidth for VoIP sessions in the Intranet Pipe
•
NAT support when the server is behind a NAT router
The Public Pipe is the WAN connection to the public Internet, including IP-VPN over WAN.
The Intranet Pipe is the enterprise WAN connection, for example, Frame Relay.
Note: The VoIP connections through public or enterprise WAN will work without configuring the IP Networks tab. However, if the total number of VoIP connections exceeds the WAN bandwidth, the voice quality will be affected for all connections. It is recommended that you set a limit based on the WAN bandwidth to ensure the voice quality.
Figure 203. The IP Networks sub-tab
Defining Your Network
If you need to configure either bandwidth control or NAT support, you have to define your network first. These are the guidelines:
•
You must define your LOCAL network IP address range. When a Pipe is defined as Local, it tells the system that the configured IP address range is not subject to bandwidth control. If the AltiGen system and this Local Network are behind the same NAT router, you need to check the Private Network check box.
This tells the system that VoIP connection to this address range does not require IP address translation, which is replacing the system's private IP address with a public address when sending VoIP packets to outside devices.
•
If you have an intranet linking multiple locations, you must enter the IP address range and define the Pipe as Intranet. If the AltiGen system and this intranet are behind the same NAT router, you need to check the Private Network check box.
•
If you have VPN service over public WAN, you must enter the VPN IP address range and define the Pipe as Public. If the AltiGen system and this VPN IP addresses are behind the same NAT router, you need to check the Private Network check box.
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Managing IP Networks
•
All undefined IP addresses fall into the Public Pipe range and are subject to bandwidth control if the public pipe bandwidth control is enabled.
Note: When MAXCS is behind a NAT router, and you do not check the Private Network check box, IP phones may not function.
Define an Address Range
1.
Click the Add button in the IP Network panel.
2.
Fill in a range of IP addresses.
3.
Select the pipe for this IP address range.
4.
If this is a private network, check the Private Network check box. Click OK.
To edit a network you’ve added, select it and click the Edit button. To remove it, select it and click the Remove button.
Configuring a Public or Intranet Pipe
If you want to regulate how many VoIP sessions can be connected to the server through a Public or Intranet
Pipe,
1.
In the Public Pipe panel, check the Enable box.
2.
Enter the maximum WAN bandwidth you want to allocate to VoIP connections.
3.
Specify the maximum sessions for one of the codecs. The system will calculate the maximum sessions for the other codecs automatically.
4.
You can change the G.711 sessions by using the Up/Down arrow button.
Figure 204. The Public Pipe options
Notes
•
When calculating the maximum sessions for each codec, the system uses the following bandwidth requirement to ensure that each session has some safety margin:
G.711 / G.722 – 90 kbps
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G.723 – 24 kbps
G.729 – 30 kbps
•
It is recommended that you use 20 ms frame size for G.711, G.722, and G.729 when configuring a Codec
Profile.
•
When different IP devices using various codecs connect to the server through a Public Pipe, the system will aggregate the total bandwidth of all connections. If the total bandwidth exceeds that specified in the
Bandwidth for VoIP box, the system will reject additional connection requests.
Configuration example
Suppose your company has a T1 line configured as half voice PRI and half data service. There are 12 remote employees using IP phones connecting to the AltiGen system. Because bandwidth is limited, you would like to regulate the bandwidth used by VoIP. You have set up remote IP phones using G.729 with 20 ms frame, and you want to limit the number of concurrent VoIP sessions to 6. If you enter 180 in the Bandwidth for VoIP field, the system will show that 6 G.729 sessions are allowed.
Configuring MAXCS Behind NAT
Your MAXCS system should be inside a firewall/NAT router. If your MAXCS is supporting remote IP phones,
IPTalk or AltiClients, you need to configure MAXCS and the NAT router to make MAXCS work properly behind
NAT. Port forwarding configuration on the firewall/NAT router is required. If you’re not sure how to configure your firewall/NAT router, please consult your firewall/NAT router manual or vendor. AltiGen Technical Support will not be able to help with this.
Important:
If your firewall/NAT router supports SIP, you need to FULLY disable this feature on the firewall/
NAT router, or conflicts may occur between MAXCS and the firewall/NAT router. In this case, remote IP phones might not work or might behave strangely. Again, please consult the firewall/NAT manual to find out how to do this.
Take the following steps:
1.
Make sure the MAXCS system uses a private static IP address, for example, 10.10.0.8. Do not use DHCP on the MAXCS system.
2.
MAXCS system is included in the range. If the range is not defined correctly, all the IP phones will not work.
Figure 205. The range of local IP addresses
3.
Set local IP network ranges to private. Multiple private networks can be added.
4.
On the IP Networks tab, in the NAT Support panel, check Enable SIP NAT support.
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Defining the IP Dialing Table
Do not check
Figure 206. The NAT Support options
Enter the Public IP address of the router in the Public IP Addresses panel. Do not check Enable Virtual
IP Addresses Support.
5.
Configure the NAT/firewall to forward TCP ports 10025, 10027, 10032, 10037, 10050, 10064, 1720 and
UDP ports 69, 5060, 10060 to MAXCS.
6.
Configure the NAT router to forward to MAXCS UDP ports 49152 + gwid * 512 ~ 49152 + (gwid * 512 + ipresno * 2) where gwid is the gateway id and ipres number is the number of the IP resource channels in the system. (See note below for an easier way to figure the port ranges.)
For the MAX1000 system, it would be UDP ports, from 49152~49211 (30 IP resource channels).
Note: An easy way to find out the RTP/TCP port range(s) for SIP is to look in the Current Resource
Statistics window in MaxAdministrator (View > Current Resource Statistics). All the ports are listed in the Local Ports column.
Implementation details
After you complete the NAT configurations, the system will translate the sending party's IP address with the defined public IP address instead of the system's private IP address. When the remote IP device sends VoIP packets to the defined public address, all packets will be routed to the system's provided IP address by the
NAT router.
Defining the IP Dialing Table
The IP Dialing Table is used for creating location-based VoIP routing in the enterprise.
To use a MAXCS-to-MAXCS connection for VoIP, you need to configure the routing in the IP Dialing Table for each MAXCS system.
Notes
•
The IP Dialing Table is disabled unless there is a VoIP board installed.
•
You must assign an IP Trunk Access code (System Configuration > Number Plan tab)
•
You must set the VoIP codec profiles.
To manage the IP dialing table, click the IP Dialing Table sub-tab in Enterprise Manager:
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Defining the IP Dialing Table
Figure 207. IP Dialing Table tab in Enterprise Manager
The left side of the window displays the VoIP domain name, the server ID length, and the name, ID and statuses of the global servers in this Domain.
To add an entry to the IP Dialing Table, click the Add button below the table (in the panel on the right).
Define the attributes for the entry:
Dialing Entry
Parameter
Server ID
Description
A unique dialing number to connect to the remote server. The server could be MAXCS, a 3rd-party VoIP gateway, or an AltiGen-certified
3rd-party VoIP device.
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The Multi-site VoIP Domain
Dialing Entry
Parameter
Server Name
Server IP Address
Remote Ext. Length
Dialing Scheme
Protocol
Codec
Hop Off Allowed
SIP Source Port
SIP Destination Port
Publish as a global entry
Description
A descriptive name of up to 15 characters to identify the server. This name may be used by Caller ID.
The remote server’s address. If the server has multiple IP addresses, enter the one that other servers will use to communicate to this system.
This IP address format is recommended over DNS names, since with the IP address, the application does not need to resolve the name.
DNS name is also posted in this field.
The length of extension digits at the remote location. Valid entries are
None – 7, with “None” meaning not specified. Specifying the remote extension length is optional but highly recommended, since this information tells the system how long to wait for another entry before sending the digits.
Overlapping (ATGN) allows the terminal to omit part of the digits required to complete a call while buffering the remaining digits. This results in faster response time, but it only works if the other end is also a MAXCS system.
Enbloc allows the system to buffer all of the digits required to complete a call.
SIP - Select if the destination supports SIP protocol.
Select which codec profile to use. If the selected profile is incompatible with the remote end, the call will not go through.
If you create two items that point to the same IP address, they must use the same codec. Specifying a different codec is an invalid configuration. MAXCS will always use the codec defined in the first item.
Choosing Yes allows calls from this remote system to hop off to the
PSTN by using the trunks in this system. Hop-off capability can be enabled or disabled on a per IP Dialing Table Location basis.
Used by UDP only. Choose the SIP source port.
Used by UDP only. Is 10060, by default.
If you are adding a system or 3rd-party VoIP device that is not part of the VoIP domain, but you want it to be seen by all servers in the domain, check this box. (The entry will appear as “Global” in the Type column.) You can also globalize it later by selecting the entry in the
IP Dialing Table and clicking the Publish as Global button below the table.
The Multi-site VoIP Domain
Note: This feature is not applicable to the MAXCS Private Cloud service.
A group of AltiGen systems can form a VoIP domain where they share the same global extension directory and call routing rules. The VoIP domain is based on VoIP framework and uses IP tie-trunks to interconnect among different sites.
A domain is created in MaxAdministrator. Here, a system is designated as the domain Master. Other AltiGen systems can then be added to a VoIP domain.
The VoIP domain Master maintains global configurations and propagates the configurations to all the members belonging to this domain automatically. Any changes in the global configuration are propagated in real time to the other members in the VoIP domain.
Note: A multi-site installation requires an AltiGen Enterprise license.
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Multi-Domain Enterprise Management Features
This section lists the PBX that are features supported under the context of multi-domain Enterprise
Management in Release 8.0.
Global Extension Features - Features supporting global extension transparently
•
Global extension Dialing (Phones and Clients)
•
Direct Dial (video not supported)
•
Conference
•
Transfer (Blind and Supervise)
•
Hold/Resume
•
System/Station/Phone Speed Dial
•
Global BLF
•
Global Extension status monitoring (Idle/Ringing/Busy status)
•
Users can call Global extensions by pressing the assigned BLF softkey
•
Call pick-up is not supported
•
Call a Global WG/Hunt Group (the agent must have a local extension)
•
View Global Paging Group
•
Global extension Forwarding/Routing
•
Trunk In Call Routing
•
DNIS Routing
•
Caller ID Routing
•
Extension forwarding options
•
Workgroup/Hunt Group forwarding options
•
Holidays routing
•
AA
•
IVR Call to Ext/Group
•
VM Record Message
•
Dial-by-Name
•
Other Call Features
•
Message Notification
•
Reminder Call
•
ONA
•
MeetMe
•
Emergency Notification
•
Return call from VM
•
Dialing from VM
•
Monitoring List
•
Maxcommunicator/MaxAgent/MaxOutlook
•
Directory
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The Multi-site VoIP Domain
•
Monitoring
•
State (Idle/Busy/Ringing)
•
Name
•
DND/Forward
•
Location
•
Department
•
Activity
•
#93 (Intercom) a Global extension
•
Advanced Call Router - call routing target
Enterprise Management Related Features
•
Global Extension User Management
•
Changing/resolving Extension/Group Scope
•
Relocate Global Extension
•
Relocate VM
•
Redirect (AltiGen phone and IPTalk)
•
#27 and #17 Global Relocation (#17 is not included in the "Relocating Global Extension for an IP
Phone" section)
•
Global LCR
•
Global DID Routing
•
Global Directory
•
Global Dial-by-Name
•
Greeting Synchronization
•
Hop-on and Hop-off calls
•
Data Transport between servers
•
ANI
•
DNIS
•
Caller Name
•
User Data
•
IVR Data
•
Tenant ID (not supported in 8.0)
•
PSTN Re-Routing
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The Multi-site VoIP Domain
Creating a Multi-site VoIP Domain
To create a multi-site VoIP domain and designate a system as the domain Master:
1.
Select VoIP > Multi-Site Domain Configuration. The Enterprise Location Manager window opens.
The name of the server appears in the Location Name field, and the name of your AltiGen product appears in the Switch Type field (MAXCS ACC or MAXCS ACM). The domain name is blank, and the server role is currently Stand-alone.
2.
Check the Allow this server to be added to domain check box.
3.
Enter a Domain Name and a Member Key.
The Member Key will be the security password when the Domain Admin adds this location into the domain. To reduce the complexity of administration, you can use the same key for all member systems.
The Enterprise Location Manager window will look similar to the next figure.
4.
Click Create domain and join as master.
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The Multi-site VoIP Domain
5.
Enter the IP address of this system. If this system has multiple IP addresses, enter the one that can communicate with other member servers.
6.
Click OK and wait for 5 to 60 seconds, depending on the size and configuration of the system. The display in the Enterprise Location Manager window changes to show the name of the VoIP domain and this server as Master.
Declaring Additional Servers for the VoIP Domain
Additional servers are added to the VoIP domain in Enterprise Manager, but first you must “declare” these servers and assign them a member key in MaxAdministrator. To do so:
1.
Log on to the member server you want to declare.
2.
Select VoIP > Multi-Site Domain Configuration. The Enterprise Location Manager window opens.
The name of the server and the name of the AltiGen product appear in the top box.
3.
Check the Allow this server to be added to domain check box.
4.
Enter the name of the VoIP domain that you want this server to be a part of.
5.
Enter a member key for this server. The Member Key is the security password when the Domain Admin adds this server into the domain. To reduce the complexity of administration, you can use the same key for all member systems.
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Working with Servers in the Domain
6.
Click Apply, then click Close.
Repeat these steps for each server you want to make available to the VoIP domain. To actually add a server
to the domain using Enterprise Manager, see “Adding a Server to a VoIP Domain” on page 346.
Working with Servers in the Domain
In the Global Server Location panel in Enterprise Manager, you can add a server to the VoIP domain by using the Add button in the panel, remove a selected server from the VoIP domain by using the Remove button, and you can set the master server, by selecting a server and clicking the Set as Master button. Before you can
Servers for the VoIP Domain” on page 344). These are the fields in the Global Server Location panel:
Server Parameter
Domain Name
Server ID Length
Global Server Location
Description
The name of the VoIP domain.
Length is from 1-3. See ““Changing the Server ID Length” on page 345 for
detailed information.
Displays the ID, Name, and Status (active/inactive) of the servers in the VoIP domain.
Master – One Domain system must be assigned as Domain Master to propagate configuration data to member MAXCS systems. The master acts as a central server to accept the connection, synchronize change from one site to the other sites, and authenticate users.
Changing the Server ID Length
The Server ID is used for the following two purposes:
•
Identifying member systems in the VoIP domain
•
Mapping to a remote system's IP address in the IP dialing table for system-to-system dialing
Depending on the number of systems that will be added to the VoIP domain and the number of entries in the
IP dialing table, the Server ID Length can be set to 1, 2, or 3 digits.
Caution!
The Server ID Length can be changed. However, if this number is changed, the server IDs are all
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Working with Servers in the Domain altered. If you increase the length, the number 0 is added to the front of the server IDs. For example, if you change the length from 2 to 3, original ID 02 and 27 will become 002 and 027 respectively. If you change the length from 3 to 2, the original IDs 112 and 311 will become 12 and
11. It is advisable to keep the original length. If you are not sure about future expansion, using a
3-digit length is advised.
Adding a Server to a VoIP Domain
Important:
Before you add a server to the domain, you need to make sure that the System ID (specified in
MaxAdministrator, System Configuration tab) is not the same as another member server’s
System ID. Enterprise Manager will use the System ID to build a unique identifier in the multisite database. Once a server is joined to a domain, you cannot change the System ID in
MaxAdministrator.
1.
Click the Add button in the Global Server Location panel.
2.
Define the attributes for the server, and click OK:
Server Parameter
Name
Address
Server ID
Member Key
Definition
Enter the name of the server.
Enter the IP address of the server.
A unique dialing number to connect to this server.
Enter this server’s member key. (Configured in this server’s Enterprise
Location Manager: VoIP > MultiSite Domain Configuration).
After you add a member server to the Domain, an entry is also added to the IP dialing table and propagated to all members automatically.
In the Global Server Location panel, the status will show "Active" if the Domain master communicates to the member successfully.
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Working with Servers in the Domain
In the event that you need to shut down the Domain Master for a period of time, you can change the Master role to another member system by selecting one of the member systems and clicking the Set as Master button.
Rejoining a Server to the VoIP Domain
If a slave server crashes, or for some other reason disconnects and never returns by itself into the domain, you will have to manually rejoin it to the VoIP domain:
1.
Rebuild the slave, if necessary.
2.
In the VoIP > Multi-Site Domain Configuration window, make sure the slave’s Server Role is Stand-
alone and that the domain name is correct.
3.
The System ID of this slave should be the same as it was before it became disconnected from the domain.
(This ID is set in MaxAdministrator: in System Configuration > General tab.)
4.
In Enterprise Manager, Global Server Location panel, select the slave and click the Rejoin button to synchronize the slave with the domain.
A dialog box opens.
5.
Input the address and member key, and click OK.
Setting an Alternate Server for AltiGen IP Phones
In a VoIP domain, you can set an alternate server to which global AltiGen IP phones will be registered when their own server (primary server) experiences a problem that interrupts phone service. The IP phones will register to the alternate server. This applies to a workgroup, as well. Switchover must be enabled for the individual
IP phones/groups in Enterprise Manager (User button > Resolve tab). But before you can do this, you must set an alternate server.
Notes:
•
Because of its role in the domain, the domain master cannot use this feature.
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Working with Servers in the Domain
•
This feature does not apply to extensions using IPTalk.
•
Make sure the alternate server has enough licenses, such as agent licenses, station licenses, and so on.
To set an alternate server,
1.
Click the Servers button, and then the Information tab.
2.
In the AltiGen IP Phone Redirect panel at the bottom of the tab, check Enable Switchover to Alternate
Server.
3.
Select an alternate server from the list.
4.
Click Apply.
(After you click Apply, the current active server name will appear in the Current Active Server box. This name is not editable.)
With the alternate server assigned, you can now configure individual extensions/groups for redirection. See
“Redirecting AltiGen IP Phones When a Server Is Down” on page 355.
Note: If the alternate server assignment is removed from the configuration above, the redirection configuration is removed from all extensions and workgroups to which you assigned this feature (User button > Resolve tab).
Note: If Native VM Integration with Microsoft Exchange is also configured, then both the primary and alternate servers need to have the same dial plan configured in the Microsoft Exchange server, so that users who have extensions flagged for redirection can access their voice messages from both the primary and alternate servers.
If the Primary or Alternate Server Is Behind NAT
When you configure the redirection feature for AltiGen IP phones, the primary server sends the IP address of the primary and alternate servers to the IP phone. The IP phone may run on the public or local network, and the primary server or alternate server may run behind NAT. So to support a server behind NAT, the primary server sends the NAT IP address or local private address according to the IP phone's IP address. If the IP phone's IP address is in a local network for the server, the primary server sends the private address, otherwise it sends the NAT address.
To configure for NAT,
1.
In Enterprise Manager, click the Servers button > IP Networks tab.
2.
In the NAT Support panel, check Enable SIP NAT Support.
3.
Configure the NAT address.
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4.
In the IP Network panel, configure the IP range of the local network or public network.
When Will Switchover Happen?
If the current active system is the primary server, switchover will happen under one of the following conditions:
•
Network error on the primary server or the primary server is down. IP phones cannot connect to the primary server. After one minute of retrying, the IP phones will register to the alternate server. At that time, the status of the primary server is "Disconnected" or "Softswitch Offline".
•
Softswitch service on the primary server is down. The status of the primary server is "Softswitch Offline".
•
IP phone service on the primary server is down. IP phones cannot register to the primary server. After one minute of retrying, the IP phones will register to the alternate server. At that time, the status of the primary server is "Fail".
•
Default gateway on the primary server is down. The status of the primary server is "Fail".
•
Manual switch from Enterprise Manager. The status of the primary server is "Standby".
When the primary server is recovered, the status is "Standby".
If the current active system is the alternate server, only clicking the Switch Back to Home Server button in Enterprise Manager can switch the control from the alternate server back to the primary server. Before manually switching back, the status of the primary server should be "Standby". After you have manually switched back, the status of the primary server changes to "Active".
Note: Unlike normal relocation, redirect can be executed only on the destination site.
Managing Domain Users
Click the User button in the toolbar to:
•
Display all extensions from all VoIP domain member systems: extension number, name, type, home server, and scope. The scope of an extension is discussed in the following section.
•
Resolve conflicting extensions and groups to global user or back to local user (on the Resolve tab).
•
Relocate an extension from one location to another location with optional voice mail (on the Resolve tab).
The General tab displays read-only information about the selected extension.
Figure 208. The User sub-tab
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When a global extension is added to a member system, this extension can be propagated to other networked systems in the VoIP domain automatically. This extension is recognized as a remote extension by other systems. When a call is made to a remote extension, it is redirected to the remote system over IP automatically.
Note: No virtual extension configuration is needed to forward the call. The Enterprise VoIP domain uses the
User directory combined with the IP dialing table to resolve multi-site routing.
An extension can call a remote extension when invoking basic features such as an extension-to-extension call, call transfer, conference, Zoomerang, and so on. Advanced features, such as silent monitoring and barge-in, between sites are NOT supported.
PSTN Failover When the TCP/IP Network is Down
Enterprise call routing works with a SIP-tie trunk, but at times the TCP/IP network may be down. To provide failover for these times, you can assign a PSTN number to each MAXCS in Enterprise Manager. The default
PSTN number is the main number of each MAXCS site.
To enable global extension rerouting,
1.
In Enterprise Manager click the Servers button, and then the Information tab.
2.
In the Global Extension Re-Routing panel, check Re-route outgoing calls when SIP tie-trunks are
unavailable.
3.
Enter a PSTN number in the PSTN Number for Re-routed Incoming Calls field, if different from the main number of the MAXCS site. If nothing is entered in this field, the main number of the MAXCS site is used.
If you enter a number, use the E.164 format.
When failover is needed, MAXCS dials the destination site number with the proper call prefix and area code or country code. On the call destination site, the call comes into the AA. The AA receives the extension number the call is directed to and rings the extension.
Note: The rerouted call may hear 1 or 2 seconds of auto-attendant announcement before the call is sent to the extension.
The Scope of an Extension in the VoIP Domain
When an extension is added to a system in MaxAdministrator, Extension Configuration window, it can be defined as Global by checking the Global extension box. If this box is not checked, the newly added extension is a local extension.
The scope of an extension shows the relationship of the extension to other member systems. In Enterprise
Manager, a selected extension’s scope appears on the Resolve tab.
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You may see any of the following in the Scope column:
•
Global – The extension has been published to all member systems within the same VoIP domain. Every extension in the domain can dial and ring this number.
•
Local – The extension has not been published to the VoIP domain. Only extensions in the same system can dial and ring this number.
•
Not Found – The extension is not a Global extension and is not created in the selected system as Local.
The extension number is used by other member systems as a local extension.
•
Remote – The word Remote in the Scope column shows that the selected system maintains this extension in the extension list because it is a Global extension of another member system. If you see an extension whose Type is Remote in the Extension Scope window, you can only see the extension information.
You cannot configure any tabs because it is created in another system.
•
Conflict – Conflict happens when one of the following situations has occurred:
•
The same extension number exists as a Global extension in one member system and as a Local extension in other systems.
•
The same extension number was created as a Global extension in different systems before the VoIP domain was formed.
The following example may help you conceptualize the multi-site extension scope.
Suppose you have three systems in different locations connected over the IP network. The numbering for
System A is 1xx; System B is 2xx, and System C is 3xx.
System A is configured as the VoIP domain Master. Assuming there is no conflict , the following table shows the Scope relationship of Global vs. Remote:
100
200
300
Ext
System A
(Domain Master)
Global
Remote
Remote
System B
Remote
Global
Remote
System C
Remote
Remote
Global
In the event that multiple systems have a same extension or group number created, the following situations may occur:
Ext
401
402
System A
Local
Local
System B
Not Found
Local
System C
Not Found
Not Found
Scope
Local
Local
Note
1
2
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Ext
403
404
System A
Global
Global
System B
Local
Global
System C
Local
Local
Scope
Conflict
Conflict
Note
3
4
1.
Extension 401 is created in System A for local purposes. Users in Systems B and C cannot dial and ring extension 401.
2.
Extension 402 is created in both Systems A and B. You may intentionally set it up this way so that System
A and B users can dial 402 for their local purposes. Ext. 402 may be used for connecting to a paging device, for example.
3.
Extension 403 is created in all systems. It is defined as Global when created in System A and not defined as Global when created in Systems B and C. This conflict requires resolution, or else System B and C users cannot dial to the Global extension in System A.
4.
Extension 404 is created in Systems A and B as Global prior to the creation of the VoIP domain. This conflict also requires resolution to determine which system will host the Global extension.
Changing an Extension’s Scope from Local to Global
If you need to resolve a conflict by making a Local extension into a Global extension, follow these steps:
1.
Select the extension in the User panel, and click the Resolve tab.
2.
Select the server name/extension where you would like the Global extension to reside.
3.
Click the Change to Global button.
Note: You must take the voice mail box and extension configuration into consideration when you change an extension to Global. In making this change, you will be deleting the voice mail box and extension settings on the home system of the “other” Local extension. A warning box will prompt you for confirmation.
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Changing an Extension’s Scope from Global to Local
If you want to change an extension's scope from Global to Local, you can highlight the extension and click the
Change All Global to Local button. This extension's scope in other member systems will be impacted after
Global is changed to Local. Using the previous case as an example, you may encounter one of the following situations when changing an extension's scope from Global to Local.
Situation 1: One Global and no conflict
Before you make the change, extension 100’s scope is as follows:
Ext.
100
System A
Global
System B
Remote
System C
Remote
After you change extension 100 to Local, the scope of 100 will be:
Ext.
100
System A
Local
System B
Not Found
System C
Not Found
Note: After you make the change, users in Systems B and C cannot dial and ring extension 100. Only System A users can call local extension 100.
Situation 2: One or more Global with conflict
Before you make the change, the scope of extensions 403 and 404 is as follows:
Ext.
403
404
System A
Global
Global
System B
Local
Global
System C
Local
Local
After you change the two extensions to Local, their scope will be:
Ext.
403
404
System A
Local
Local
System B
Local
Local
System C
Local
Local
Note: After you make the change, extensions 403 and 404 can be dialed only by the users in their own system.
Relocating a Global Extension for an IP Phone
The administrator can relocate a global extension from one system to another. In addition, a user may be allowed to relocate a global extension by using the feature code #27. To allow a user to use this feature, check the appropriate check box in the Relocation panel on the Resolve tab. The behavior of this feature differs, de-
In Release 8.0, Polycom users can also relocate extensions. Refer to the MAXCS Polycom Configuration
Guide
for requirements and instructions.
Note: The check box is available only if a global extension is selected and that extension has no conflict.
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User can relocate extension, if checked
Admin can always relocate extension
When a global extension (extension 1001, in this example) is moved from site A to site B, this is what happens:
•
The following configurations are replicated from site A to site B:
•
First Name
•
Last Name
•
Password
•
Extension Number
•
DID Number
•
Dial by Name
•
Disable Mailbox option (Extension Configuration, Mail Management tab)
•
Site A marks extension 1001 as removed and adds it to a Relocated Extension List (REL). The configuration of extension 1001 is still remembered in site A, even though it appears to be removed.
•
Site B creates extension 1001. If extension 1001 is found in site B’s REL, the extension 1001 will be restored in site B. However, the fields listed above will be overwritten with the settings of site A’s extension
1001. If extension 1001 is not found in site B’s REL, a new extension 1001 will be created in site B. The fields listed above will be set with site A’s extension 1001 settings. The remaining fields of extension 1001 in site B are set with default values.
For the administrator to relocate a global extension,
1.
Select the extension in the User list. The Relocation panel shows where the extension is located.
2.
From the To box, select a different system for the extension.
3.
To move the extension’s voice mail along with the extension, check the Relocate VM check box. Then select either Relocate VM Now or Relocate VM after x hour(s).
Note: Because moving the voice mail requires network bandwidth, you may want it to move when system usage is low. The first time the voice mail is moved to a specific location, it can take hours for all the voice mails to be moved. Thereafter, only new voice mails are moved (because the old ones are still there, backed up), so subsequent moves take a shorter time.
VM files are transferred by HTTP protocol using TCP port 10043. The administrator can configure the firewall/router to limit the bandwidth on port 10043, so that the voice mail transferring will not impact the voice quality over IP.
Note: If you do not move the voice mail, the VM files will be deleted and cannot be recovered. (When the
user
relocates an extension using #27, the voice mail is moved also. The user cannot choose whether or not to move the voice mail.)
4.
Click Relocate.
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Notes on Relocating a Global Extension
•
The phone user can start using the voice mail during VM relocation, but the voice mail count will keep increasing until the relocation is complete.
•
If extension 1001 is relocated from site A to site B, and the administrator creates a local extension 1001 in site A, the extension 1001 will be removed from the REL. Later, if the administrator removes the local extension 1001 and relocates global extension 1001 back to site A, this extension cannot be restored to its original settings.
•
When an extension is relocated to site B for the first time, the administrator or the user should configure the Call Restriction, Speed Dial list, and so on, for one time in site B. These configurations will be stored on site B. Later, if the extension is relocated to site B again, no additional configuration is needed, as the previous configuration will be restored.
•
If multiple systems in the VoIP domain have a PRI interface, it’s possible that DID numbers could be duplicated. For example, say the DID number for extension 1001 is configured as 250. In this case, the
DID number 5102520250 and 4087899250 will ring extension 1001. To ensure that this doesn’t happen, you can do one of two things: (1) Make sure the DID numbers are not duplicated; (2) Ask the CO to send more digits (to decrease the likelihood of identical DID numbers).
Relocating a Global Extension Using #27 on Analog Phone vs AltiGen IP Phone
•
Analog phone: The phone must be off hook. The user presses #27 and follows the voice prompts. User must press # after inputting the password.
•
IP phone: The IP phone must be on hook. The user presses #27, and then inputs the global extension number and password. The global extension is then relocated to this IP phone.
If system B does not have a prior record of this extension, it will create a new extension with known information and the following settings:
•
Enable IP Extension and Dynamic IP Address settings will be selected automatically (in MaxAdministrator, Extension Configuration window).
•
The newly created extension will use the default voice mail, mail forwarding, notification, call handling, restriction, and monitor list settings (MaxAdministrator, Extension Configuration window).
Note: The administrator needs to make the proper changes for this user when the global extension is relocated by the user.
•
When this Global extension user returns to his home office, all settings are stored in the REL database.
The administrator does not need to change these settings when the user presses #27 to relocate the extension the next time.
Relocating More Than One Global Extension
When more than one global extension is being relocated at the same time, and voice mail is also being relocated, the voice mail of the extension that was relocated first will be copied over completely to the relocation site, before copying begins for the voice mail of the second extension, and so on.
The extension, itself, is relocated immediately.
Redirecting AltiGen IP Phones When a Server Is Down
Relocating a global extension, described in the preceding section, is intended to serve employees who are physically relocating to another office for a time. Administrators can also configure global IP phones to register to another server in the VoIP domain when their primary server goes down for some reason. All configured phones switch over at the same time. When their primary server returns to service, the administrator can switch the phones back to their primary server by clicking the Switch Back to Home Server button in the Servers >
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Notes:
•
When you redirect AltiGen IP phones, voice mail is not moved. Otherwise, the extension configuration changes of the redirect feature are the same as they are with normal relocation.
•
If Native VM Integration with Microsoft Exchange is also configured, users can access their voice messages from both the primary and alternate servers, if both have the same dial plan configured in the
Microsoft Exchange server.
•
Redirection does not work when an extension user is using IPTalk.
Before configuring individual IP phones to redirect from their primary server to an alternate server, an alternate server must be assigned in Servers > Information tab > AltiGen IP Phone Redirect panel. The Redirect option is not available until an alternate server is assigned. Only AltiGen IP phones that are global and have no conflict with the extensions of other sites can be configured to redirect.
To configure an IP phone to redirect,
1.
In Enterprise Manager click Users button > Resolve tab.
2.
Select a global IP phone whose server has an alternate server assigned.
Note: The AltiGen IP extension may need to be pre-configured on the alternate server to match its configuration on the home server, so that it works as expected. (For example, the alternate server may have a different call restriction policy. The extension on the alternate server may belong to a different workgroup. The greetings may be different even if the extension number is the same.)
3.
Check the Enable Switchover to Alternate Server check box.
Note: If an extension configured with the redirection feature is manually relocated (by the system administrator in Enterprise Manager or by the user pressing #27), the redirection configuration is dropped on the new site. If the extension is manually relocated back to its original site, the feature is recovered.
Changes to AltiGen IP Phone When Redirect Is Configured
After the redirection feature is configured, the IP phone will receive the configuration of the primary and alternate server address, and store them in its local flash memory. Once it has been configured for redirection, the IP phone’s “AW Server” address will be that of the primary server. The user can view the address on the
IP phone (Menu > System > AW Server) but cannot configure it. When redirect is enabled, "Primary Server" and "Alternate Server" are added to the phone’s System menu. They are read-only.
IPTalk Redirect
The Redirect feature, if enabled , will prompt users to connect to a backup server if their client application loses its connection with the primary server.
This feature has been implemented in a similar fashion in the following MAXCS client applications:
•
MaxAgent
•
MaxCommunicator
•
MaxOutlook
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Enable IPTalk Redirect
The Redirect feature, if enabled, will prompt users to connect to a backup server if their client application loses its connection with the primary server. This has been implemented in MaxAgent, MaxCommunicator, and
MaxOutlook.
To enable this feature, Administrators perform two steps:
1.
Enable the Redirect feature for MAXCS.
2.
Enable the Redirect ability for specific user extensions.
To enable the Redirect feature ,
1.
In MAXCS Administrator, enable the Redirect feature for MAXCS. Select VoIP > Enterprise Network
Management.
2.
Click the Servers button on the Toolbar. Select the Information subtab.
3.
Check the option Enable Redirection to Alternate Server.
4.
For Alternate Server, select the server that you want to use as the alternate server. Click Apply.
5.
Next, enable the Redirect feature for specific users. In Enterprise Manager, click the User button on the
Toolbar.
6.
Select the Resolve subtab.
7.
Select the users who will be able to use this switchover ability.
8.
Select Enable Switchover to Alternate Server. Click Apply.
Figure 209. Enable switchover for a user
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9.
Assign the IPTalk seat license to the remote extension to enable the IPTalk Redirect feature. You do this in the Client Seat License Management window.
Switch Users Back to the Main Server
In the event that the main server goes down or is otherwise unavailable, you can ask users to switch client applications from the alternate server back to the main server by following these steps:
1.
Open Enterprise Manager, click Servers on the Toolbar, and select the Information subtab.
2.
Click Switch Back to Home Server.
MaxAgent and MaxCommunicator Switchback Behavior
For MaxCommunicator and MaxAgent, when you enable this feature for MAXCS, users that have been assigned the switchover ability will experience the following behavior if their client loses its connection with the
MAXCS server.
1.
When the client application detects that its connection with the MAXCS server has been lost, the client will present a pop-up message to the user. The message will inform the user that the connection has been lost, and will offer the user several options.
Figure 210. The client application prompts the user to reconnect or switch servers
2.
If the user chooses Reconnect to Main Server, then the client will try to re-establish its connection with the main server. If it cannot reconnect, it will prompt the user to redirect the connection to the alternate server.
If the user chooses Redirect to Alternate Server, then the client will establish a connection with the alternate server.
3.
Once the Administrator has brought the main server back up and clicked Switch Back to Home Server within Enterprise Manager, the user will be prompted to switch the client's connection back to the main server.
MaxOutlook Switchback Behavior
Users that have been assigned the switchover ability will experience the following behavior if MaxOutlook loses its connection with the MAXCS server.
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1.
When the connection with the main server is lost, a message opens.
Figure 211. MaxOutlook prompts the user to close and restart the application
2.
After the user closes Outlook and restarts it (within 5 minutes), MaxOutlook prompts the user to either reconnect or redirect.
Figure 212. MaxOutlook prompts the user to either reconnect or redirect
3.
If the user does not restart MaxOutlook within 5 minutes, MaxOutlook will not redirect, and the user will see the login page showing the login server address as the main server.
Once you have brought the main server back up and clicked Switch Back to Home Server within Enterprise
Manager, the user will be prompted to switch the client's connection back to the main server.
Configuring Departments in a Multi-site VoIP Domain
In a VoIP domain, departments can be defined and added to extensions. An extension in one MAXCS system can be assigned to only one department. However, the same extension number in different MAXCS systems can be assigned to different departments. A department can also be assigned to a global extension and can be seen across the Enterprise domain.
In MaxAdministrator, the department field can be seen on the Extension General tab. In MaxCommunicator, the department is displayed on the Directory and Monitor tabs. In Enterprise Manager, the department is displayed in the User list. Departments can also be seen in CDR Search.
To define a department and assign or remove members from a department, click the Department button.
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Figure 213. Department configuration
To define a department,
1.
Click the Add button at the bottom of the Department panel.
2.
Enter a department name and a description, if desired, and click OK.
To configure extensions for departments,
1.
Select a department in the Department list.
2.
To add non-member extensions to the department, select the extensions and click Add.
3.
To delete extensions from the Member Extensions list, select the extensions, and click Remove. To remove all member extensions from a department, click Remove All.
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Configuring Global Least Cost Routing
Global LCR allows you to save on toll charges by making long distance or international calls through a VoIP domain member system. The target system will function like a PSTN gateway for other member systems to hop-off. For example, suppose you have two systems in the U.S. and one system in the U.K. configured as
VoIP domains. When users in the U.S. dial country code 44, you want the call to be dialed though the system in the U.K. to its PSTN network.
Global LCR has higher priority than local outcall routing. The system will check the Global LCR entries first before the call is handled by the local system's outcall routing rules.
Before you configure Global LCR, you need to evaluate the following conditions:
•
How many concurrent calls will be routed through the target system?
•
Does the target system have enough PSTN trunks to support the entire VoIP domain?
•
Does the target system have enough WAN bandwidth to support system-to-system and PSTN hop-off calls?
Before you configure Global LCR, you need to make sure the following settings are properly configured in Max-
Administrator:
•
Both systems need to have the route access code configured on the Number Plan tab in System
Configuration. (The user has to dial the route access code + the phone number to use Global LCR.)
•
The target system needs to have the hop-off restriction reference properly configured. The reference extension is set on the Call Restriction tab in System Configuration, and then that reference extension cannot have Internal Calls Only checked on the Restriction tab of Extension Configuration.
To configure Global Least Cost routing,
1.
Click the Global LCR Button.
2.
On the Global LCR screen, click the Add button.
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3.
Fill in the dialog box, and click OK.
Parameter
Enable
E.164 Number Pattern
Calling From
Transit Through
Description
Check this check box to enable the configuration.
E.164 is the ITU standard format for international telephone numbers.
Enter a country code and area code. For example, the number pattern for a site in Fremont, Calif., would be 1510 (the country code 1, followed by the Fremont area code 510).
Select the server from which the call originates, or select
All Servers
.
Select the server that receives the call.
4.
After adding a route, click Edit, check the Enable check box, and click OK to activate the Global LCR route.
To edit an entry made to the Global Least Cost Routing table, select the entry you want to change, and click the Edit button. Make your changes, and click OK.
When Information May Be Out of Sync
If a server is down for any length of time, such that changes may have been made in the VoIP domain and the server is now out of sync with the Master, you need to update the server manually. In the server’s
MaxAdministrator, select VoIP > Multi-Site Domain Configuration, and click the Replicate from Domain button. This brings the server up-to-date with the Master.
If the server is still not seeing all the information it should (this would be rare), click VoIP > Refresh Enterprise
Configuration.
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H A P T E R
26
System Report Management
MAXCS provides a System Summary report, an IP Cumulative Traffic Statistics report, and an SNMP (Simple
Network Management Protocol) configuration screen, all available from the Report menu.
System Summary Report
The System Summary report provides summary information on extensions, trunks, and workgroups configured in the system. To open the System Summary report window, select Report > System Summary, or click the
Summary button on the toolbar.
Figure 214. System Summary window
•
Extension Summary – Configured extensions in the system, including Extension number, Last Name,
First Name, SMTP/POP3 E-mail name, Slot (Logical board ID), and Channel.
•
Group Summary – Configured workgroups and hunt groups in the system. When you select a group, agents belonging to that group are displayed in the Member window.
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•
Trunk Summary – Configured trunks in the system, including trunk location (Board ID : Channel Number) and trunk access code assignment.
•
Messaging Usage – Message count and storage usage for each mail box. Click the Refresh button to update the message count and storage size information.
You can print this report by clicking the Print button.
IP Cumulative Traffic Statistics
To view a report of all cumulative IP traffic, click Reports > IP Traffic Statistics. The window shows IP trunk traffic information for all calls:
Figure 215. IP Cumulative Traffic Statistics window
This window displays the following data:
Cumulative
Traffic Field
Internet Address
Packets Sent
Packets Received
Bytes Sent
Bytes Received
Packets Lost
Average Jitter
Description
The IP address of the VoIP system or device.
Number of voice packets sent to other systems over the public or private IP network.
Number of voice packets received from other systems over the public or private IP network.
Total size (in bytes) of all voice packets sent to other systems over the public or private
IP network.
Total size (in bytes) of all voice packets received from other systems over the public or private IP network.
Number of voice packets that have been lost due to prolonged delays, network congestion, or routing failure.
Average length of delay per voice packet in milliseconds. This figure should stay under
100 milliseconds. A higher figure indicates a longer average delay. This number can be used to measure the quality of service on the network that connects the source and destination sites.
The difference between the Current Resource Statistics window and the IP Cumulative Traffic Statistics window is that the former shows figures only for the active call (Current Traffic) on a particular IP trunk of the local MAXCS system while the other window shows figures for all calls combined (cumulative traffic).
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Resetting Cumulative Statistics
You can reset the IP Cumulative Traffic Statistics by clicking the Reset button. Also, this window automatically resets all fields to 0 when the MAXCS system is shut down and restarted. Statistics gathered before the reset are not saved.
Using SNMP
SNMP (Simple Network Management Protocol) is used in network management systems to monitor networkattached devices for conditions that warrant administrative attention.
This MAXCS SNMP configuration, used with a third-party management console (see the next section), helps you see the MAXCS status, so you can use MAXCS more securely. Using an SNMPv3 agent, MAXCS sends
SNMP traps to the management console when alarming conditions are detected.
Note: SNMP traps are sent by the AltiGen services SPServ (Softswitch up, Softswitch down traps), AltiKeep
(warm start trap), and AltiServ® services (all other traps), so those services must be started, or the traps will not be sent.
SNMP traps are logged in text files: SNMP_Altiserv.txt, SNMP_DinaProxy.txt, SNMP_GlobalAppearance.txt, and SNMP_SPServ.txt.
SNMP Management Console
To use SNMP, you need an SNMP management console that is SNMPv3-supported for receiving and collection. AltiGen recommends MG-Soft Trap Ringer Professional Edition, available from MGSoft
Corporation, at http://www.mg-soft.com/tringer.html.
You can get help about how to configure an SNMP User Account and Management Console Port in that product’s Help system.
Note: AltiGen’s IANA Private Enterprise Number is 13679.
Configuring MAXCS for SNMP
To configure MAXCS for SNMP, select Report > SNMP Configuration.
Configure the parameters:
•
Check Enable sending SNMP traps.
•
Enter the SNMPv3 server address.
•
Enter the SNMPv3 server port.
•
Select a security level:
–
No Authentication and No Privacy
–
With Authentication but No Privacy
–
With Authentication and Privacy
•
Select an Authentication method and enter a password
•
Select a Privacy protocol and enter a password.
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Configure traps:
•
Memory usage exceeds – This trap is sent when MAXCS detects that the lowest virtual memory usage exceeds a specified percentage of physical memory configuration within a 10-minute duration. Default value is 80%. The next trap will be sent after the condition is cleared then occurs again. The minimum duration between any two consecutive traps is 30 minutes
•
Average CPU utilization exceeds – This trap is sent when MAXCS detects its average CPU utilization exceeds a specified percentage in any 10-minute duration. Default value is 80%. The next trap will be sent after the condition is cleared then occurs again. The minimum duration between any two consecutive traps is 30 minutes
•
Hard disk usage exceeds – This trap is sent when hard disk usage of MAXCS transitioning from below threshold to on or above threshold is detected. Default value is 80%. The minimum duration between any two consecutive traps is 30 minutes.
List of Traps Sent
A trap is sent when the following conditions are detected.
•
Cold Start (generic trap). When AltiServ is cold started and initialized successfully.
•
Warm Start (generic trap). When AltiKeep service detects AltiServ.exe is down, re-starting AltiServ.exe, and AltiServ is initialized successfully.
•
LinkDown (generic trap). When detecting a T1/E1/PRI span state is transitioning from up to down or losing clock source.
When a gateway is down, one trap is sent for each T1/E1/PRI interface in this Gateway. This trap is sent when SIP trunk destination state transitioning from reachable to unreachable is detected.
Every T1/E1/PRI span and SIP trunk channel is assigned a unique "ifIndex" value as a port identifier
•
LinkUp (generic trap). When a T1/E1/PRI span state transitioning from down to up is detected.
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•
Softswitch up (specific trap). When AltiServ.exe starts to respond to the keep alive packets sent by the
SNMP Agent. AltiServ should respond to the keep-alive packets after its initialization is completed.
•
Softswitch down (specific trap). When AltiServ.exe stops responding to the keep-alive packets sent by the
SNMP Agent.
•
Gateway/Media Server connection up (specific trap). When a gateway or HMCP Media Server connection state transitioning from down to up is detected.
•
Gateway/Media Server connection down (specific trap). When a gateway or HMCP Media Server connection state transitioning from up to down is detected.
•
Enterprise Manager Master up (specific trap). When MAXCS is in Enterprise Manager slave role and Enterprise Manager master state transitioning from down to up is detected.
•
Enterprise Manager Master down (specific trap). When MAXCS is in Enterprise Manager slave role and
Enterprise Manager master state transitioning from up to down is detected.
•
Enterprise Manager Slave up (specific trap). When MAXCS is in Enterprise Manager master role and detects Enterprise Manager slave state transitioning from down to up.
•
Enterprise Manager Slave down (specific trap). When MAXCS is in Enterprise Manager master role and detects Enterprise Manager slave state transitioning from up to down.
•
IP Phone service up (specific trap). When detecting IP Phone service transitioning from down to up.
•
IP Phone service down (specific trap). When detecting IP Phone service transitioning from up to down.
•
VM server connection up (specific trap). When detecting VM server connection transitioning from down to up.
•
VM server connection down (specific trap). When detecting VM server connection transitioning from up to down.
•
CT Proxy Service up (specific trap). When CTProxy Service connection transitioning from down to up is detected.
•
CT Proxy Service down (specific trap). When detecting CTProxy Service connection transitioning from up to down.
•
Excessive memory usage on Softswitch (specific trap). When MAXCS detects the lowest virtual memory usage exceeds a specified percentage of physical memory configuration within a 10-minute duration. The next trap will be sent after the condition is cleared then occurs again. The minimum duration between any two consecutive traps is 30 minutes.
•
Excessive CPU utilization on Softswitch (specific trap). When MAXCS detects its average CPU utilization exceeds a specified percentage in any 10-minute duration. The next trap will be sent after the condition is cleared then occurs again. The minimum duration between any two consecutive traps is 30 minutes.
•
Excessive hard disk usage on Softswitch (specific trap). When hard disk usage of MAXCS transitioning from below threshold to on or above threshold is detected. The minimum duration between any two consecutive traps is 30 minutes.
•
SIP Trunk Group Status traps (Trunk Group Up, Trunk Group Down, Trunk Group Busy)
•
SIP Abnormality traps (Out of HMCP codec resources, insufficient SIP extension channels, malicious IP address added)
•
Security traps (Toll fraud - outbound call from voicemail exceeding call duration threshold or call quality threshold, 911 call, extension lockout)
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•
Out of licenses traps (Station Session, ACM Agent Seat [work as session], Recording seat, Recording session, MaxCommunicator seat, MaxCall seat, MaxAgent seat, MaxSupervisor seat, MaxMobile seat,
SDK Connection session, Trunk Control APC SDK session, IPTalk seat, HMCP G.711/G.723/G.729 VPR,
HMCP MeetMe Conference, HMCP Agent Supervision session, Integration Connector seat, Exchange
Integration seat, MaxInsight session, MaxCommunicator session, MaxAgent session, MaxSupervisor session, AltiConsole session, IPTalk session, MaxCall session, 3rd Party SIP Device seat)
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H A P T E R
27
Microsoft Exchange Integration
This chapter provides step-by-step instructions for configuring Microsoft Exchange and MAX Communication
Server (MAXCS) to work together.
An AltiGen Exchange Integration license is required for each extension using Exchange integration.
Note: Note that MaxOutlook does not support the 64-bit version of Microsoft Outlook; only the 32-bit version is supported.
description of these options):
•
Synchronize with Exchange – Synchronize voice messages between the AltiGen voice mailbox and
Exchange mailbox. Works with Exchange 2007, 2010, and 2013
•
Bridged Access to Exchange – An option is provided in the AltiGen Voice Mail System menu to log in to the Exchange mailbox (option 7 in the main menu). To synchronize voice mail between the AltiGen mail box and the Exchange server, check the Enable Synchronization check box. If you don’t check this, voice mail is not synchronized between the two message stores.
•
Native VM Integration with Exchange – In this mode, the AltiGen voice mailboxes are replaced by
Exchange mailboxes. Each user in MAXCS needs to have a mailbox in the Exchange server and each mailbox must be Unified Messaging (UM) enabled, or the user will not be able to receive any voice mail.
You can choose any of the three options while installing MAXCS, and later you can switch options from
MaxAdministrator (in the Voice Mail Configuration window). If you upgraded from AltiWare 5.2 and you were using Exchange integration, your configurations are kept and the option is set to Synchronize with Exchange.
When you switch options, service restart is required.
Note: If Exchange integration was used in the previous version of MAXCS server, you must uninstall the
Outlook client before you install MAXCS 8.0.
MAXCS Private Cloud Service
Exchange integration requires MAXCS to be a member of the same domain as the CPE network. Therefore, forwarding voicemail to email should be used instead of Exchange Integration.
Requirements
Make sure the following items are ready before Exchange integration is configured. Note that AltiGen is not responsible for, and cannot support, installation of Microsoft Exchange Server:
To set up any kind of Exchange integration, you need the following:
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•
One Windows server for MAXCS, loaded with:
–
Windows Server or Windows 7
–
The MAX Communication Server ACM software
•
A second Windows server for Exchange, loaded with Exchange Server software. It should be installed on
64-bit system(s) with Windows 64-bit or above OS. Unified Messaging, Client Access, and Mailbox Server roles should be installed with Exchange Server.
Important: When you install both the Exchange Server and MAXCS, you must log in as the Domain
Administrator, NOT the Local Administrator.
•
The MAXCS system and the Exchange Server system must belong to the same domain, with a network throughput rate of no less than 100Mbps and without any Web proxies in between.
•
AltiGen Services must be installed and started with the user account
<Domainname.com>\AltiGen_<AltiServSystemName> or
<AltiServSystemName>
@<Domainname.com>. Make sure that you include the .com or .net part of the domain name in the entry.
This service account must have a mailbox in the Exchange Server that is different from the previous version.
•
Exchange Server Services must be started.
•
Successful ping from Exchange Server to MAXCS and vice versa.
When You Install MAXCS
You may be installing MAXCS now, or you may have already installed it. To integrate with Exchange, you need
MAXCS 8.0.
Note: The installation program will check whether Microsoft Outlook has been installed on the server. If it detects Outlook, you must uninstall Outlook before you can install MAXCS.
1.
If you are installing now, log in to Windows with a user account that is a member of the Domain Admin
group. If MAXCS is already installed, skip to step 4.
2.
you have a chance to change the default password. Record this password for future troubleshooting.
Figure 216. In this example, the MAXCS installation program created a user account
“AltiGen_LESLIEXIA” in the domain VMDOMAIN. We changed the password to “altigen”.
3.
After installation, add this user account to the Domain Admin group via Active Directory Users and
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4.
If MAXCS was already installed on the system, do the following: a.
Create a new domain user account, and add it to the Domain Admin group via Active
Directory Users and Computers. b.
Move the MAXCS server to the Domain.
c.
Use the AltiPassword change utility (C:\AltiServ\Exe\AltiPwdChange.exe) to change all
AltiGen service accounts to run as this new user account.
Note: In the future, if you need to debug you must log in to the MAXCS server with this user account.
Figure 217. Add the user created by the MAXCS install program (or created by you in step 4) to the
Domain Admin group in Active Directory Users and Computers.
Exchange 2010 Integration Configuration Steps
This section describes how to configure integration with Exchange 2010. For instructions on configuring
MAXCS with Exchange 2013, see Exchange 2013 Integration Configuration Steps on page 384.
After installation, perform the following steps.
1.
Add Exchange Integration licenses to MAXCS (see Figure 218).
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Figure 218. Adding Exchange Integration licenses in MAXCS
2.
In the Exchange Management Console, create a mailbox for the service account that was created during
installation (or created by you in step 4, above) (see Figure 219).
Figure 219. Creating a mailbox for the service account created during installation
3.
In MaxAdministrator, choose System > Voice Mail Configuration, then select the Exchange Integration
mode you want to use, and enter the name (not the IP address) of Exchange server (see Figure 220).
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Select the Exchange
Integration mode you are going to use.
Enter the NAME (not the IP address) of the
Exchange Server
Figure 220. Choosing the Synchronize Exchange Integration mode in MAXCS
4.
Configure the names of each extension user such that the first and last names are the same as the user’s matching mailbox on the Exchange Server.
Note: The Middle Initial field should be empty for Exchange Server mail accounts in order for Exchange integration to work properly.
5.
MAXCS matches the mailbox on the Exchange Server via the display name, which is a combination of
make sure the user’s display name on the Exchange Server is “Michael Wang”, or synchronization will fail.
Figure 221. First name and last name in MAXCS must match the display name in Exchange Server, or synchronization will fail. (Also, to synchronize MAXCS voice mail with Exchange voice mail in Bridged Access mode, you must have checked the Enable Synchronization check box in the Voice Mail configuration screen.)
Note: Exchange Native VM integration uses the extension number and the extension’s first and last names to link between MAXCS and Exchange.
Exchange synchronization uses only the Extension’s first and last names as the link.
6.
For users whose voice mails will be integrated with Exchange, check Assign Exchange Integration
License on the Extension Configuration screen’s Mail Management tab (see Figure 222). Make sure that
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Figure 222. Assigning the Exchange Integration license to a user
Note: Each user needs to access their mailbox once via an e-mail client (Outlook, Outlook Express, Outlook
Web Access) before synchronization will start working for that user.
This is all you need to do if you selected the Synchronize with Exchange option in the Voice Mail
Extra Steps for Bridged Access and Native VM Integration
In addition to the steps given thus far in this chapter, follow these additional steps for Bridged Access and
Native VM integrations.
1.
Create a dial plan in Exchange. In the Exchange Management Console, go to Organization
Configuration > Unified Messaging, and click New UM Dial Plan.
The digit length you enter must match the digit length of extensions in
MAXCS
Figure 223. Creating a new UM dial plan. In this example, the name of the dial plan is “LeslieXiaSS”.
2.
Enter a name for the dial plan and length of extension numbers. The digit length must be the same as the extension number length in MAXCS. Then click New.
When you create a dial plan, a default UM Mailbox Policy is created automatically and associated with the
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Figure 224. A Default UM Mailbox Policy is created when you create a new dial plan
3.
After creating a dial plan, open its Properties dialog box, select the Settings tab, and change Audio
Codec to G.711 (see Figure 225).
4.
Click OK.
Figure 225. Changing Audio Codec to G711
5.
Add your MAXCS server as a UM Gateway: Go to Organization Configuration > Unified Messaging >
UM IP Gateways > New UM IP Gateway.
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Figure 226. The name of our example gateway is “LeslieXia” and the name of the dial plan we created is “LeslieXiaSS”.
a.
Enter the name of the gateway.
b.
Enter the IP address of your MAXCS server.
c.
Browse for and select the dial plan you just created.
d.
Click New.
6.
If your system has multiple gateways, repeat step 5 to add all of your gateways as UM IP Gateways.
7.
Associate your dial plan to the Exchange Server UM. To do this, in the Exchange Management Console, go to Server Configuration > Unified Messaging, select the server and click Properties.
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Figure 227. We added our new dial plan, named “LeslieXiaSS”, to the list of associated dial plans
8.
Click the UM Settings tab, click Add, and add your dial plan to the list of associated dial plans.
This completes all system-wide settings in Exchange Server.
Configuring UM Settings for Each User
With all system-wide settings in Exchange Server complete, configure the UM settings for each user.
1.
In Recipient Configuration > Mailbox, select the user and select Enable Unified Messaging from the
Actions pane.
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Figure 228. Here, Leslie Xia is an individual IP phone user with a mailbox in VMDOMAIN.
2.
Click Browse, select the policy associated with the dialing plan you created, then click OK.
Figure 229. The policy associated with the dialing plan we just created is “LeslieXiaSS Default Policy”.
3.
Enter the user’s MAXCS extension number in the Manually entered mailbox extension field (make sure the extension number is the same in MaxAdministrator and the Exchange User Mailbox).
4.
Select PIN setting(s), and click Next. (If you select Automatically Generate, the Exchange Server will send the user an e-mail with the PIN.)
5.
Click Enable.
6.
Repeat steps 1-5 for each user you want to enable.
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Configuring Out Calling from UM
This section shows how to enable extensions integrated with Exchange in Native or Bridged mode, to do the following tasks:
•
Call a personal contact or a contact from the database
•
Return a call from Exchange voice mail
Note: Unlike with AltiGen’s Zoomerang feature, a user calling out from voice mail cannot go back to the
Exchange voice mailbox after returning the call.
Configure the following in Microsoft Exchange.
1.
Check Allow outgoing calls through this UM IP gateway.
Figure 230. Allowing outgoing calls through the UM IP gateway
2.
Set the Dial Code in your dial plan:
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Figure 231. Setting the Dial Code
Outgoing Configuration
•
Outside line access code – The trunk access code of your Softswitch
•
International access code – Toll call prefix for international calls. For the U.S., it is "011"
•
National number prefix – Toll call prefix for domestic calls, always set as "1"
•
Country/Region code – Country code. For the U.S., it is "1"
Incoming Configuration
•
In-country/region number format
•
Use this field to specify how a user's telephone number should be dialed by the UM Server in a different dial plan, but having the same country code. This is used by an auto attendant and when an Outlook
Voice Access subscriber searches and tries to call the user in the directory.
•
This entry consists of a number prefix and n number of x characters (for example, 020xxxxxxx).
•
To determine the telephone number, UM will append the last n-digits from the telephone number that is specified in the directory to the prefix that is specified.
•
International number format
•
Use this field to specify how a user's telephone number should be dialed by the UM Server in a different dial plan, and having a different country code. This is used by an auto attendant and when an Outlook
Voice Access subscriber searches and tries to call the user in the directory.
•
This entry consists of a number prefix and n number of x characters (for example, 4420xxxxxxx).
•
To determine the telephone number, UM will append the last n-digits from the telephone number that is specified in the directory to the prefix that is specified.
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3.
On the Dialing Rule Groups tab, add dial rules for in-country/region and international calls that will be placed by UM-enabled users. Each dialing rule entry that is defined on the dial rule group determines the types of calls that users within a specific dial rule group can make.
Figure 232. Adding Dial Rule entries in the Dialing Rule Groups tab
For a Dialing Rule Entry (see (see Figure 232)), the following are required:
Name – Select a name of an existing dialing rule or, if you want to create a dialing rule, type the name of the dialing rule (up to 32 characters, text characters only). This is the display name for the dialing rule that will be displayed in the Exchange Management Console.
Number Mask – Define the number mask for the dialing rule. A number mask is used to define the telephone number format that a Unified Messaging server will use to determine what outgoing telephone number it will dial for a user. When an outgoing call is made to a number that is matched by the number mask on the dialing rule, the UM server will substitute the digits that are matched into the dialed number. It will then use the digit string from this match to make the outgoing call. An example of a valid number mask is
91425xxxxxxx. This field can contain only numbers and the letter 'x'.
Dialed Number – Define the dialed number for the dialing rule. The dialed number is used to determine the actual dial string that is sent to the IP gateway. This number can be different from the number that is obtained by Unified Messaging for the outgoing call. However, your PBX can also be configured to omit the area code for local calls and can be configured for private voice numbering plans. Any wildcard (x) characters in the dial string are substituted with the digits from the original number that were matched by the number mask on the dialing rule. An example valid dialed number is 9xxxxxxx. This field can contain only numbers and the character "x".
Comment – Use this text box to input a comment or description for the dialing rule that you are adding.
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Figure 233. Creating a dialing rule
For example, if the business number of a personal contact is 5102529712, then the number mask should be set as "91510xxxxxxx", because UM will add "91" automatically, and the Dialed Number is "9xxxxxxx", so that the final dialed number will be "92529712".
You can use the wild card "*" to handle any length of digits.
4.
Assign the Dial Entry to mailbox Policies: Go to UM Mailbox Policies, select the mailbox that users belong to, open the Dialing Restrictions tab, and assign the rule group you just created.
Figure 234. Assigning dial plan to mailbox policy
After you configure the UM mailbox to use a dialing rule group, the dialing restrictions that are configured apply to all UM-enabled users who are associated with the UM mailbox policy. For example, you can configure a dialing rule group that does not require users who are associated with the dial plan to dial an outside line access code when they place a call to an in-country/region telephone number.
Note: If you need help in configuring dialing rules, see http://technet.microsoft.com/en-us/library/ bb629580.aspx. That will put you in the general location of what you need. Much of this information came from that Microsoft site.
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Configuring MaxAdministrator
Complete the configuration in MaxAdministrator.
1.
Go to System > System Configuration > Number Plan tab. In the First Digit Assignment panel, assign one of the digits (for example, digit 8) to IP Trunk Access.
2.
(Click the first SIP-Tie entry, and assign the digit, then use the Apply button to apply the assignment to all the other SIP-Tie entries.) This allows calls in either bridged or native mode to access the Exchange
Server.
3.
Go to VoIP > Enterprise Network Management to open Enterprise Manager.
4.
Click Codec to create a new codec profile only for the Exchange connection.
a.
In the Name field, enter a name for the new codec profile.
b.
In the Codec field, select G.711 Mu-Law.
c.
In the DTMF Delivery field, select RFC2833 d.
In the SIP Early Media field, select Enable.
e.
In the SIP Transport field, select TCP.
5.
Associate this new codec profile to the IP address of Exchange Server (and only Exchange Server): a.
Click the Servers button, then click the IP Codec tab.
b.
Add a new IP Device Range for the Exchange Server:
Figure 235. Associating your “Exchange” codec profile to the IP address of Exchange Server
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6.
Click the Add button in the IP Device Range panel.
7.
Select the codec profile you just created specifically for Exchange.
8.
Enter the IP address of the Exchange Server in both the From and To fields. Be sure that this IP address does not fall into any other device range. (Check the IP Codec tab and the IP Dialing Table tab.) If it does, reset that range into two ranges: one that ends just before the Exchange Server’s IP address, and one that starts just after the Exchange Server’s IP address.
9.
In MaxAdministrator, go to System > Voice Mail Configuration. In the Microsoft Exchange Integration panel, select Bridged Access to Exchange or Native VM Integration with Exchange. Click OK.
10. Restart all AltiGen services.
Note: After all AltiGen services are restarted, voice mail access may be unavailable for 1-2 minutes.
Exchange 2013 Integration Configuration Steps
This section describes how to configure MAXCS integration with Exchange 2013.
1.
Add Exchange Integration licenses to MAXCS.
2.
In the Exchange Management console, create a mailbox for the service account that was created during installation (choose recipients > mailboxes, click the Plus sign and choose user mailbox).
Figure 236. Add a new user mailbox
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Figure 237. Specify details for this user mailbox
3.
In MAXCS Administrator, select System > Voice Mail Configuration. Select the Exchange Integration mode that you want to use, and enter the name (not the IP address) of the Exchange server.
Figure 238. The
MAXCS
Administrator Voice Mail Configuration panel
4.
Configure the names of each extension user such that the first and last names are the same as the user's matching mailbox on the Exchange Server. Leave the Middle Initial field empty.
5.
In order to synchronize voicemail in Bridged Access to Exchange 2007 mode, check the Enable
Synchronization
option below that option.
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6.
For users whose voicemail messages will be integrated with Exchange, open the Mail Management tab for the user's extension and check the option Assign Exchange Integration License.
Figure 239. The “Assign Exchange Integration License” option for an extension
Bridged Access and Native Voicemail Integration
1.
Create a dial plan: In the Exchange Management console, select unified messaging > UM dial plans.
Click the Plus sign.
Figure 240. Create a new dial plan
2.
Enter a name for the dial plan, specify the length of the extension numbers, and enter a country or region code. The extension length must be the same as the extension number length in MAXCS.
A default UM Mailbox policy is automatically created and is associated with this dial plan.
3.
In the UM dial plan list, select the dial plan that you just created and click the pencil icon to edit the dial plan.
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Figure 241. Click the pencil to edit the dial plan
4.
Click Configure.
Exchange 2013 Integration Configuration Steps
Figure 242. Click the Configure button
5.
Select settings and set the Audio Codec option to G.711. Save and close the windows.
Figure 243. Set “Audio Codec” to G.711
6.
Select unified messaging > UM IP gateways. Click the Plus sign.
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7.
Enter the name of the gateway, the IP address of the MAXCS server, and browse to select the dial plan you just created.
If your system has multiple gateways, repeat this step to add each gateway.
Figure 244. Create the gateway(s)
8.
Select servers > servers. Select the name of the Exchange server and click the Pencil icon to edit the settings. (In the next figure, the name of the Exchange server is EXCH.)
Figure 245. Click the Pencil icon to edit the exchange server
9.
You will add the new dial plan to the list of associated dial plans. Select unified messaging. Below
Associated dial plans, click the Plus sign.
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Figure 246. Click the Plus icon to add the new dial plan to the list
10. Select the new dial plan and click add. Click OK.
Configure UM Settings for Each User
1.
Select recipients > mailboxes. Select the user and click the Pencil icon.
2.
Select mailbox features. Below Phone and Voice Features, click Enable.
Figure 247. Enable Unified Messaging
3.
Browse to the policy associated with the dialing plan that you just created. Click Ok. Click Next.
4.
Enter the extension number for the user to access the mailbox through Outlook Voice Access, and specify the PIN options. Click Finish.
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Figure 248. Enter the extension number and PIN
Repeat these steps for each user.
Configuring Out Call Routing from UM
This process enables extensions that are integrated with Exchange in either Native or Bridged mode to call personal contacts or contacts from the database and to return calls from an Exchange voicemail message.
5.
Select unified messaging > UM IP gateways.
6.
Select the gateway. Click the Pencil icon.
7.
Select the option Allow outgoing calls through this UM IP gateway.
8.
Select unified messaging > UM dial plans. Click the dial plan and click the Pencil icon.
9.
Click configure. Select dial codes.
Figure 249. Open the Dial Codes panel
country/region and international calls that will be placed by your UM-enabled users.
11. Complete the steps in the section Configuring MaxAdministrator on page 383.
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When You Create a New Mailbox User
When You Create a New Mailbox User
If you are using Synchronize mode, Bridged Access mode with synchronization, or Native VM Integration mode, and you create a new mailbox user in Exchange Server and a new extension in MAXCS, to associate them you need to restart the AltiGen Exchange Integration Service.
Testing for Synchronization
You can use some simple procedures to make sure that the Synchronize with Exchange integration is working correctly.
To test the integration, set up an extension in MAXCS (for example, extension 100) and its corresponding mailbox in Exchange Server. Also, set up a computer with Outlook configured for this user.
To test message delivery to Exchange
1.
Leave a voice mail for extension 100. The message light illuminates.
2.
Log on to the Exchange Mailbox from Outlook and check for the message in the inbox. The message should be titled Voice-mail from xxx and include the voice mail as a
.wav
attachment.
To check message state change notification
1.
Log in to extension 100’s voice mail from a phone. The message you left in the preceding step should be there as a new message.
2.
Save the message by pressing 3. Within approximately a minute, the message in Outlook will become a saved message as well – it will no longer appear in bold.
To listen to VM in Outlook
Open the message in Outlook, and open the
.wav
attachment. It should be the same message.
To check deletion notification
1.
Delete this message from Outlook.
2.
Wait a few minutes, and then log on to extension 100’s voice mail from a phone. The voice mail should no longer be there.
If any of these tests fail, consult the “Troubleshooting Tips” section.
Troubleshooting Tips
To check the profile for the service account
1.
Log on to the MAXCS system as the AltiGen service account (for example, AltiGen_telesystem). You will need the password you set up when you installed MAXCS.
2.
Select Control Panel > Mail. (In Windows, right-click Microsoft Office on the Start menu, and select
Properties.)
3.
Click Show Profiles. In MAXCS 8.0, there is only one profile there, which is for the service account, so that name should be AltiExch<ServerName><AccountName> (for example,
AltiExchMAILSERVERAltiGen_telesystem).
If you don’t see any such profile, make sure that \altiserv\exe folder does not contain the files
mapi32.dll
or
gapi32.dll
. If these files exist, delete them, then stop and start the Exchange
Integration Service.
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To delete the profile for the service account
If an error occurred while MAXCS was creating the service account profile, the damaged profile would remain there until removed manually. After the re-configuration, the new profile can't be created, because the old one still exists.
You can remedy this in the following way:
1.
Log on as AltiGen Service Account.
2.
Shut down AltiGen Exchange Integration Service from Control Panel > Administrative Tools >
Services, then open Control Panel > Mail (or Mail and Fax) and click Show Profiles. Remove the service profile.
3.
Start the AltiGen Exchange Integration Service from Control Panel > Administrative Tools >
Services.
If this doesn’t work, contact AltiGen Technical Support.
To gather trace files
1.
Log in to Admin.
2.
Select Turn AltiTrace On, and click Apply.
3.
Select VM and SP Log Dump.
4.
To view logs, go to AltiServ\Log\VM\ExchIntg.
To avoid “extension in use” message
When synchronizing with Exchange Server, the mailbox needs to be locked. If the extension has a lot of messages, it could take some time, but shouldn’t take as long as 2-3 minutes. In normal cases, it should take just
10-20 seconds. You may adjust a registry key to change the synchronization interval:
HKEY_LOCAL_MACHINE\SOFTWARE\AltiGen Communications, Inc.\AltiWare\ExchIntg\Polling Interval
The value is in ms. 60000 = 60 seconds. You may change it to 300000 for 5 minutes. After changing the value, restart Exchange integration service for the change to take effect.
Exchange Integration service synchronizes voice messages on the Exchange server with those on the MAXCS system by polling the two servers periodically. This polling interval can be adjusted by creating a DWORD value called "Polling Interval" under the key
HKEY_LOCAL_MACHINE\SOFTWARE\AltiGen Communications, Inc.\AltiWare\ExchIntg
This DWORD value should contain the number of milliseconds between polling. If this value is not present in the registry, a default value of 60000 (1 minute) is used by the system. For performance reasons, you should not set this value to below 60000.
To avoid “Access Deny” errors while sending messages
If you have applied Microsoft patch ms06-029, when an AltiGen PBX phone user attempts to send a message, the user receives an "Access Deny" error. This is because the patch changes the grant for the permission of
Send As.
After applying the patch, the Send As permission of each user needs to be granted to the account of “altigen service” explicitly.
You may have to restart the Exchange Server and MAXCS.
Notes
•
Prevent attempts by the Exchange Administrator/Manager to use the existing service account for the
AltiGen Exchange Integration Service. Using the AltiGen service account will provide you an audit trail that is invaluable while troubleshooting.
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•
Depending on the number of voice mails you have on the AltiGen server, the initial mailbox synchronization may take a long time.
For example, if you have 10GB of voice mails on the AltiGen server and are enabling Exchange integration for all the mailboxes, it may take up to 24 hours to initialize the Exchange integration service.
On the other hand, if you have less than 100MB of voice mails on the AltiGen server, the initialization will take less than 5 minutes.
•
If users experience a problem making calls to the Exchange server, make sure the MSXML 6.0 Parser has not been deleted from the server. Without it, the speech engine services cannot play voice prompts.
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Fax-over-IP Configuration
This section describes how to configure MAXCS for Fax-over-IP (FoIP).
MAXCS supports FoIP, leveraging T-38 pass-through. (T.38 pass-through is only supported on AltiGen SIP
Trunks.)
Both voice calls and fax calls can be made through a FoIP extension; both incoming and outgoing calls on the extension will use G.711 codec.
Fax extension-to-extension is supported; the SIP endpoints must be a supported ATA device.
Limited FoIP extension parameters are supported:
•
First/Last Name, Password, Description, Department
•
DID Number, Transmitted Caller ID, E911 CID
•
IP Extension parameters, except Enable Fallback to Mobile Extension
•
Out Call Restrictions
•
Forward All Calls - forwarding target must be an FoIP extension or FoIP hunt group
•
Busy Call Handling - forwarding target must be an FoIP extension or FoIP hunt group
Limitations
Sending and receiving faxes over IP service has known limitations. AltiGen, along with many other companies, uses the T.38 industry standard for FoIP configuration. The T.38 standard contains minor variations in how it can be implemented. Because of these variations, one provider’s FoIP handling can vary from another’s, thus introducing the possibility of incompatibilities. As the standard continues to evolve, it is reasonable to expect these variations to diminish over time.
If your organization typically sends frequent faxes that are lengthy multiple page documents, consider retaining a few analog lines and traditional fax machines as a backup option.
Requirements
In order to implement FoIP, your system must meet the following requirements:
•
Your system must be running MAXCS Release 7.5 or later.
•
Your SIP Service provider must be AltiGen.
•
You FoIP gateway must be an AltiGen-certified ATA device. Supported devices include various
AudioCodes gateways; see the AltiGen Knowledge Base (https://know.altigen.com/) for device configuration guides.
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Overview
MAXCS supports T.38 pass-through, which is a protocol for real-time fax transmission over IP networks. There are three common scenarios for sending faxes to/from MAXCS.
One scenario is an On-premise installation of MAXCS with users sending faxes within the facility.
Figure 250. Example of internal fax configuration in an on-premise MAXCS deployment
In this scenario, faxes are connected to VoIP gateways via analog FXS. The gateways all connect to MAXCS via network connections with T.38 pass-through.
Another scenario is an On-premise installation with users sending faxes to, and receiving faxes from, an external fax machine outside of the organization.
In this scenario, the faxes connect to the VoIP gateway, which connects to MAXCS. MAXCS connects to a router which sends faxes through (and receives faxes from) the cloud. The remote location will have a router receiving the fax, sending it to the VoIP gateway which is connected to the target fax machine.
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Figure 251. Example of sending faxes from internal fax devices to external fax machines
A third scenario illustrates fax flow with MAXCS Cloud deployments. The gateway connects to the cloud, connecting to MAXCS, and faxes are sent through to the router, gateway, and fax device at an external location.
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Figure 252. Example of sending and receiving faxes in a MAXCS Cloud deployment
Configure an IP Extension for FoIP
To configure an IP extension to support FoIP,
1.
Select PBX > Extension Configuration.
2.
Select the extension on the left.
3.
On the right, in the IP Extension group, check Enable Fax-Over-IP.
Figure 253. Configuring Fax-over-IP for an IP extension
Enabling this setting changes the configuration for this extension as follows:
•
All outbound calls from this IP extension will use the G.711 codec
•
All incoming DID routing calls to this extension will use the G.711 codec
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Configure In Call Routing for FoIP
MAXCS supports support two different In Call routing methods for fax numbers:
•
DNIS In Call Routing
•
FoIP Extension DID number
Routing supports T.38 UDPTL (UDP packets) using the same RTP port range.
To enable FoIP DNIS In Call routing,
1.
Select PBX > In Call Routing Configuration.
2.
Select the DNIS number.
3.
On the DNIS Routing tab, check Enable Fax-Over-IP and click OK.
Figure 254. The Enable Fax-Over-IP checkbox
Enabling this setting forces incoming calls that match this entry to use G.711 codec. If this checkbox is not checked but the destination is a fax extension, then G.711 codec will still be used.
Create a SIP Group for Fax Routing
You can allow voice and fax calls to run on the same SIP trunk channel. The trunks must be configured to support both voice and fax. The SIP trunk uses the same SIP server IP address, but different authentication credentials for voice trunk versus a fax trunk.
To configure fax routing,
1.
Log into the MAXCS Cloud Services portal and retrieve the Fax details on the General tab of your account
You will need this information in steps 6 and 7.
2.
In MAXCS Administrator, open Trunk view.
3.
Double-click an unconfigured SIP Trunk. In the Trunk Configuration panel, click Trunk Properties.
4.
In the next dialog box, click SIP Group Configuration.
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Figure 255. The SIP Groups and SIP Servers lists
5.
Add a new SIP Group: a.
Click Add (just below the Groups list) to add a new SIP Group: b.
Enter a name (for example, FoIP) and check the option Fax Trunk Routing. c.
Enter the fax trunk's user name and password (from the Fax SIP Trunk Group section on the General tab of the order in the Cloud Services portal) and click OK.
Figure 256. Adding a new SIP Group for FoIP
6.
Add two servers to this new SIP Group: a.
In the Groups list, select the SIP Group that you just created.
b.
To create the first server, click the Add button that is just below the SIP Servers list.
c.
Copy and paste the first domain (from the Fax SIP Trunk Group section on the General tab of the order in the Cloud Services portal) and click OK.
d.
Repeat this process to create the second server; click Add ( below the SIP Servers list), paste the second domain, and click OK.
7.
You now have two SIP servers in this SIP Group. Next, configure these two servers: a.
In the SIP Servers list, highlight the first server that you added.
b.
Specify the following parameters on the Register tab:
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•
For SIP Server IP Address, copy and paste the IP address of the first trunk server (from the Fax SIP Trunk Group section on the General tab of the order in the Cloud Services portal (for example, 65.254.44.194)
•
For User Name, copy and paste the fax username from the portal
•
For Password, copy and paste the fax password (click Hide/Show Password to see it)
•
Set the SIP Register Period to 60
•
Set the SIP Source Port to 5060
•
Set the SIP Destination Port to 5060
Details for the first Fax
SIP Trunk
Details for the second
Fax SIP Trunk
Figure 257. Fax SIP Trunk details on the General tab of the order in the Cloud Services portal
c.
Switch to the Settings tab and specify the following parameters:
•
Set the SIP Protocol Field to FROM Header.
•
Select Carrier can only accept assigned numbers as Calling Number.
•
Enter the range of DID numbers that you want to assign to this trunk in the Calling
Number can be accepted by the Carrier box. These must be valid DIDs that are on the trunk; each entry must be 10 digits.
•
For the Use this Calling Number if the Carrier cannot accept configured numbers field, enter the main phone number. The phone number that you enter must be included among the Calling Number can be accepted by the Carrier entries.
•
Select Send Caller Name. (Do NOT select Enable Standard Record-Route Header.)
•
Set the Incoming DID Number field to To Header.
d.
Repeat steps a through c for the second server, but this time specify the IP address as the
second
SIP trunk address from the portal.
8.
Assign channels to the SIP Group and enable those channels: a.
Close the SIP Group Configuration window to return to the SIP Signaling Channel
Configuration
window.
b.
Click Channel Assignment.
c.
Select the channels to apply to this fax SIP Group. (You can use Ctrl-click to select multiple channels.) d.
Click Assign Group. In the list, select the group that you just created and click OK.
e.
In the channel list, check the checkboxes for those channels to enable them. Click OK.
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Considerations:
•
Outbound calls that are made through SIP channels that have been configured for fax channels are for fax only. Therefore, they should not be assigned trunk access codes or be included in the out call routing for voice calls.
•
Do not add these two SIP Trunk servers to any out call routing tables
•
Do not assign trunk access codes to these two fax SIP Trunk servers
•
These two fax SIP Trunk servers can receive either voice or fax trunk calls
Configure Gateway Devices
MAXCS works with several gateways, including the following devices:
•
AudioCodes MP-202
•
AudioCodes MP-118 and MP-124
For instructions on configuring these devices, and any other new devices that may have been certified after this manual was released, search the AltiGen Knowledge Base (https://know.altigen.com/) for articles on your specific hardware.
FoIP Hunt Groups
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Tools and Applications
MAXCS comes with the following tools and applications for testing, diagnosing and configuring your system.
They are available from the Windows Start menu: Start > All Programs > MAXCS ACC/ACM:
Under Gateway Tools:
•
AltiGen Board Test
•
CT-Bus Test Tool (formerly MVIP Test Tool)
•
Gateway Configuration Tool (For information on this tool, see “Gateway Configuration Tool” on page 79.)
Under Utilities:
•
MAXCS Backup and Restore Utility
•
MaxAdministrator and Extension Security Checker
•
Start and Stop All AltiGen Services
•
Trace Collector
•
Voice File Converter
•
Read Config
In addition, on the Services > Utilities menu in MaxAdministrator:
•
Work/Hunt Group Converter utility
•
Export and Import extensions utilities
If you installed AltiGen’s Custom Phrase Manager, it is available off the Start > All Programs menu. You can use this tool only if you have an AltiGen SDK license.
AltiGen Board Test
This is an AltiGen hardware test tool for system hang and other hardware-related problems. It tests the following on all AltiGen boards:
•
Board memory from host or from both host and DSP
•
DSP internal memory from host or from both host and DSP
•
FMIC connection and data memory from host
•
NVRAM from host
•
PMC chip from host and DSP if T1/E1 board
You have the option of testing a single board or testing all boards at the same time.
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CT-Bus Test Tool
The CT-Bus Test Tool is a tool that detects one-way connection, cross talk, bad MVIP cable and static noise problems.
To run the CT-Bus test tool,
1.
Stop AltiGen Switching Services before running this utility.
2.
Launch CT-Bus Test Tool from Start > All Programs > MAXCS > Gateway Tools > CT-Bus Test Tool.
3.
Click Start to begin the test.
4.
At the end of the test, the utility provides pass or fail results.
Backup and Restore Utility
Note: The configuration backup option is turned on by default.
Backups do not include CDRs.
To back up or restore data, select either
•
From MaxAdministrator: Services > Utilities > Backup and Restore, or
•
From the Windows Start menu: All Programs > MAXCS > Utilities > MAXCS ACC/ACM Backup and
Restore.
Figure 258. Backup and Restore window
Note: The Backup and Restore window can only be accessed at the primary MAXCS system; it is not available from a remote MaxAdministrator client.
Backing Up Files
Note: The AltiGen backup utility does not back up internal CDR database. You must back up the internal
CDR database from \AltiDB\InternalDB\Internal.mdb manually.
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1.
Select the Backup icon.
Figure 259. Backup Configuration dialog box
2.
In the Components panel, select the files you want to back up.
3.
In the Backup To list, select the day of the week (each day has its own folder in C:\altibackup for backing up files to), or select Advanced to change the drive or select a different folder.
Selecting Advanced displays a folder icon. Click the folder icon to open a browse dialog box that lets you select the folder to back up to. When you click OK in the dialog box, the selected drive or directory is displayed in the field below the Backup To list.
4.
Click OK to start the backup. In the Backup and Restore window, the progress of the backup appears.
Scheduling Backups
You can set up automated backup on a schedule, and you can select the days, the times, and the target drives and folders for the backups.
To set backup schedules,
1.
In the Backup and Restore window, select the Schedule button.
Figure 260. Backup Schedules dialog box
2.
Set the options:
•
Check the box for each day of the week you want run the backup.
•
For each day, use the lists to specify the time. These time settings use a 24-hour clock.
•
You can accept the default target directories, or you can click the Folder icon to open the Browse for
Folder dialog box to select the destination for the backup files.
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•
Under Backup Selection, select the file components you want to back up: Configuration files, Custom
Phrases, Extension Messages, SP Configuration files.
3.
Click OK.
Restoring Backed up Files
To restore backed up files,
1.
Stop the AltiGen switching services.
2.
In the Backup and Restore window, select the Restore icon.
Figure 261. Restore Configuration dialog box
3.
Under Components, select the file groups you want to restore.
4.
Using the Restore From list, select the day you want to restore from, or select Advanced to choose the restore folder.
Clicking Advanced displays a folder icon that you can click to open a dialog box that allows you to select the directory you want to restore from.
Select a day of the week or manually choose the restore directory. The specified directory appears in the text box below the list.
Note: The components you select for restore must have been backed up into the directory you selected.
For example, if you didn’t back up configuration files on Thursday, you won’t be able to restore them from the Thursday directory.
Important:
Make sure the version you restore the database files from is compatible with the current MAXCS version. If incompatible files are restored, the phone system will fail to restart!
5.
Click OK to start the restore process.
6.
When you are finished restoring backed up files, restart the AltiGen switching services.
MAXCS Admin & Extension Security Checker
MAXCS Admin & Extension Security Checker is a tool that
•
Checks the security status of every extension in your MAXCS system and displays the security characteristics of each extension. From an extension’s right-click menu, you can lock and unlock the extension, force the user to change the password, clear an attacked record, and reset the status.
•
Shows how many MAXCS Admins are currently connected to the system. By clicking Disconnect All, you can disconnect all MAXCS Admins from the local MAXCS system.
Launch the MaxAdministrator & Extension Security Checker from Start > All Programs > MAX
Communication Server > Utilities > MAXCS Admin & Extension Security Checker.
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Number of Admin connections
MAXCS Admin & Extension Security Checker
Automatically refreshes the display
Information on the selected extension
Security characteristics to check
Figure 262. MaxAdministrator & Extension Security Checker
Checking Extension Security
Generally, an extension is considered secure if its password meets the following conditions:
•
Contains 4-8 digits
•
Is different from the extension
•
Is different from the default system password
•
Does not consist of consecutive numbers
•
Does not consist of a repetition of the same digit
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To check extension security,
1.
Select the security characteristics you want to check in the Show field group.
Status
Secure Pwd + Internal Only
Unsecure Pwd
Outbound-capable
Unsecure Pwd + Outbound
Password Expired
Attacked
Locked
Password Match
Description
Has secure password and cannot make outbound trunk calls
Password has unsecure elements described in the Unsecure
Elements window
Can make outbound trunk calls
Password has unsecure elements described in Unsecure Elements window AND can make outbound trunk calls
Password is expired
8 consecutive false password attempts have been made
Extension has been locked by system due to attack or by System
Administrator
To detect if an extension uses a specific trivial password, such as street address, zip code, phone number, enter that string here.
2.
Click Refresh. Extensions with the selected insecure characteristics will appear in the Extension List.
3.
Make changes to extensions from the right-click menus, or advise extension user(s) to make changes.
4.
After changes have been made (for example in MaxAdministrator, MaxCommunicator, or with right-click commands in this tool), click Reload to fetch the new settings from MAXCS.
Security characteristics for extensions you select in the Extension List display in the Unsecure Elements panel.
5.
(Optional) Click Export to export the data in the Extension List to a text file.
Note: You are advised to run this security check periodically and remind extension users to use secure passwords.
Start & Stop All AltiGen Services
You can start or stop all AltiGen services from the Windows Start menu: All Programs > MAXCS > Utilities
> Start & Stop All AltiGen Services.
To shut down all AltiGen services, click the Shutdown All AltiGen Services button. Some examples of when you might want to do this are before you upgrade, before running some utilities and tools, and to apply certain configuration changes.
To start all AltiGen services, click the Start All AltiGen Services button.
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Trace Collector
The Trace Collector collects trace for diagnostic purposes, and lets you upload the results to AltiGen Technical
Support directly from the Trace Collector dialog box. You can run the Trace Collector tool from the Windows
Start menu, and also from MaxAdministrator’s Diagnostic menu.
Note: Trace Collector is not available from an MaxAdministrator installed in a remote machine.
The Trace Collector first examines the running status of MAXCS and the gateway, and then checks whether each trace status is on or off. If a trace status is turned off, the AltiGen system will not produce those traces.
A message box pops up if MAXCS and the gateway are not running or an important trace status is off.
Parameters for Trace Collector
Customize traces by choosing from the following parameters:
Time Period for Extension Feature #66 – Defines how many hours you want to go back to collect trace, starting from the time you press #66. The default value is 2 hours.
Case Number – Enter the AltiGen case number associated with this trace collection activity. The case number will comprise the first part of the file name of the collected trace package.
Problem Description – Enter a description of the problem, including the extension number involved, the time when the problem happened, how to reproduce the problem, and so on.
Time of Incident – The tool collects the trace between the time ranges. The range covers before and after the defined Date and Time. The default Date and Time is one hour before the current date and time, and the default variation is 60 minutes. This setting is not applicable when #66 is performed.
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Trace Category – By default, all options are selected.
•
Main MAXCS Trace (\AltiServ\log)
Collects the following files, and extracts the trace records that fall in the specified time range: actrace.log
ALPxxx.txt
altiserv.txt
AltiBack_XXX.trc
AltiKeep_XXX.trc
AnnouceRunLog.txt
AssertLog.txt
AW_AstrCpyErrLog.txt
CallQManLog.txt
CDRLogDLL.txt
CDRLogTrace.txt
ConfigLog.txt
MsgOCLog.txt
MSRunLog.txt mviperr.txt
Postman.txt
ProcInfoLog.txt
AlpErrLog.txt
\atps\threadID.txtl
\atps\cmdlog.txt
AdvQOverflow.log
Ac2AppPathHdlTbl.txt
FeatServ.txt
DbUpdateTrdLog.txt
HGwGenLog.txt
HGwMsgLog.txt
threadid.txt
MidNightLog.txt
\logservice\Internal.txt
ConfigServiceLog.txt
CDRLogTrace.txt
CDRLogDLL_EXCEPTION.txt
CSH323log.txt
ExceptionLog.txt
SIPlog.txt
SIPMan.txt
SIPPstnReg.txt
SipExtChanTbl.log
SIPKeepALive.txt
QESLLog.txt
Qtmlog.txt
Loggservice_Mutex.txt
MEMORYTRACE.txt
NewCDRExt.txt
TritonSPLog.txt pathlog.txt rsrclog.txt
RtpPortRangeTbl.txt
StartupLog.txt
Swxx_xxxx.txt
GWMsgLog.txt
•
System Configuration Data – Collects system configuration data, including System, Extension, Trunk,
AA configurations, and Read OE files.
•
Service Provider Log Dump – Runs SPDump.exe to dump the SP log into files and then collects the trace.
•
IP Phone Dump – Collects the AltiGen IP Phone dump log in \AltiServ\Log\IPP (except for IP-805 models).
•
Stand-alone Gateway Trace – Collects the trace on the stand-alone gateway machine. If AltiServ
Services are shut down, the option is disabled. If Trace Collector is running on the stand-alone gateway machine, this option is hidden (because Trace Collector just needs to collect the trace locally).
•
Triton DSP Trace Dump – Collects the Triton DSP dump log in \AltiServ\SP\Triton\. Runs
TritionDSPDebug.exe to dump Triton DSP binary log data, runs TATraceDecode.exe to convert binary log to text files, and then collects the text files.
•
AltiConnect Trace Dump – Runs acdump.exe to dump the AltiConnect Trace, and then collects the trace. If Trace Collector is running on the stand-alone gateway machine, this option is hidden.
•
Windows Event log – Extracts the system and application event log from the Windows system.
Start Collecting – Click this button to begin the trace collection, according to the time range and trace categories you chose. All collected files will be zipped to a single file, which will be listed in the Collected Trace
Packages list box. The progress bar will display the progress of the whole process.
Storage Folder – The collected trace package is saved in this folder. The format of the file name is
CaseNumber_Year_Month_Day_Hour_Minute_Second _ComputerName.zip. If the trace package is collected by #66, the format of the file name is #66_Year_ Month_Day_Hour_Minute_Second _ExtensionNumber.zip.
Free Space – Displays the free space of the drive where the storage folder is located. The folder must be in a local drive.
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Change Storage Folder – Opens a folder browser window to select another storage folder. After the change,
Storage Folder, Free Space, and the package list are refreshed to reflect the status of the new storage folder.
Explore Storage Folder – Opens the storage folder in a new explorer window.
Upload Package to FTP – Opens an FTP configuration dialog box. After you complete the required configuration, Trace Collector uploads the selected package to the AltiGen Tech Support FTP site.
Apply Configurations to #66 – Apply time period, trace category, and storage folder to feature code #66
(Trace Collecting).
Limitations
If you run Trace Collector on MAXCS installed machine, note the following limitations:
•
If MAXCS is not running, Trace Collector can only collect the trace of this machine. The traces in memory, such as "AltiConnect Trace Dump" and "Service Provide Log Dump", and "Stand-alone Gateway Trace" will not be collected.
•
If the default gateway is not running, the traces for "Triton DSP Trace Dump" will not be collected.
•
If AltiServ is running, and an attached remote gateway is not running, or a remote gateway is detached, the trace for this gateway will not be collected even if that "Stand-alone Gateway Trace" is selected. If an attached gateway has the status of "Out of Service", the trace for this gateway will be collected.
If you run Trace Collector on a gateway MAXCS ACC/ACM-installed machine, note the following limitations:
•
It can only collect the trace of this machine.
•
If the stand-alone gateway is not running, the trace for "Triton DSP Trace Dump" will not be collected.
Network Log for AltiGen IP Phones
The AltiGen Network log shows traffic between the IP phones and the server. The information in this log is helpful when troubleshooting voice quality issues.
To view the log, from MaxAdministrator select Diagnostic > AltiGen Network Log.
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Figure 263. AltiGen Network Log
Voice File Converter
This tool converts phrase, greeting, and music files from .wav to AltiGen format and vice versa. To open the tool, from the Windows Start menu, select All Programs > Utilities > Voice File Converter.
Note: The source .wav file must be in 8k/8bit/mono/mu-law format.
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Read Config
You can sort by clicking a column head
To use the Voice File Converter,
1.
Beside the From field, click the Browse button to select the folder that contains the files you want to convert.
2.
Beside the Convert To field, click the Browse button to select the destination folder for the converted files.
If they are prompts, they should be placed in the C:\PostOffice\phrases\LangCustom directory on the gateway that is running MAXCS. If the files are music files, they should be placed in the
C:\PostOffice\Phrases\Music directory. A file that you want to use for music on hold must be named
MusicOnWaiting. To save the AltiGen system MusicOnWaiting file, rename it before replacing it.
3.
Check the files you want to convert.
4.
In the Format panel, select a format.
5.
Click Convert.
If a file format is incorrect, an error message pops up.
Read Config
Read Config (or Configuration Reader) is a tool that creates a subdirectory in \altiserv\EXE\AltiWareHtml\ of
HTML files showing details of your MAXCS configuration.
To use Configuration Reader,
1.
Launch Configuration Reader from Start > All Programs > MAXCS > Utilities > Read Config.
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Open previous
ReadOE file
Create new
ReadOE file
Output all configuration to this folder
Output configuration to altigen_rc.dat
Click View to see your latest HTML file
2.
Make selections in the dialog box. If you will be sending a configuration file to AltiGen Technical Support, check ReadOE Data File, and select a folder for the .dat file.
3.
Click Go.
A processing bar indicates the progress of configuration reading.
4.
When the status window is complete, you can click the View button to view the HTML files showing your configuration.
Columns across the top of the opening page let you view statistics on different components of your configuration.
Work/Hunt Group Converter
The MAXCS Work/Hunt Group Converter allows you to convert workgroups to hunt groups or hunt groups to workgroups.
To launch the Work/Hunt Group Converter, select Services > Utilities > Convert Work/Hunt Group.
Figure 264. Work/Hunt Group Converter
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Workgroups and hunt groups are listed on the left side of the window, member agents and non-agents are listed on the right side.
The Work/Hunt Group Converter can be used to convert:
•
Agent to Non-Agent – If an agent belongs to any workgroup, it cannot be converted to a non-agent. When an agent is converted to a non-agent, all workgroup-related parameters will be cleared, including wrap-up time, inter-call delay, and outgoing workgroup number.
To convert, select the agent (indicated by YES in the Agent field) and click the Convert Agent button or double-click the agent.
•
Non-Agent to Agent – To convert, select the non-agent (indicated by NO in the Agent field) and click the
Convert Agent button or double-click the non-agent. Make sure you have enough agent licenses.
•
Convert Workgroup to Hunt Group – When a workgroup is converted to a hunt group, its members are not changed, but the following parameters are cleared, including:
– voice recording setting
– queue time threshold
– queue overflow settings
– queue announcement
– agent announcement
– queue quit forward (returns to default value – to voice mail)
– call distribution (if previously configured to Ring First Available Member, Ring Next Member or Ring All, the setting is not changed. If configured to any other settings, the setting is configured to Ring First Available Member.)
To convert, select the workgroup (indicated in the Type field) and click the Convert Group button or double-click the workgroup.
•
Convert Hunt group to Workgroup – A hunt group cannot be converted if it contains at least one nonagent. You must first change the extension from non-agent to agent (by selecting the agent and clicking the Convert Agent button or by checking the Agent check box in the Extension Configuration window before converting).
To convert, select the hunt group (indicated in the Type field) and click the Convert Group button or double-click the hunt group.
Exporting and Importing Extensions
You can import and export extensions in a .csv file and you can import extensions from the active directory.
Importing Extensions from a .csv File
1.
First, back up your system configurations, using AltiGen’s Backup and Restore tool (Services > Utilities
> Backup and Restore).
2.
Go to Services > Utilities > Import Extensions.
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3.
In the Import Extensions dialog box, click the Explore button to select a .csv file to import, and click OK.
All the extension records in the .csv file are added to the Import Extensions list.
4.
Check the records you want to import. Click the Select All and Clear All buttons to select or clear all the check boxes.)
5.
Click Import.
A progress bar lets you see the progress of the import. When the import is finished, a message lets you know how many extensions were imported, how many extensions were skipped and how many extensions failed.
6.
If an extension already exists, you are prompted whether to replace the extension:
If you overwrite an extension, fields that are not specified in the .csv file are not overwritten with default values or blank values. For example, if the column Department is not included in the .csv file, but is configured in the extension that you overwrote, the Department field is not reset to the default value when the extension is overwritten.
When the import is finished, a report file opens showing detailed information for every extension you attempted to import. If some fields are invalid, the system replaces them with a default value, except for the extension number field.
416 MAXCS Administration Manual
Exporting and Importing Extensions
The name of the text file is the same as the .csv file, except that the file extension is .txt.
Importing Extensions from the Active Directory
1.
First, back up your system configurations, using the Backup and Restore tool (Services > Utilities >
Backup and Restore).
2.
Go to Services > Utilities > Import Extensions from Active Directory.
3.
Enter the server path, user name and password.
4.
Click the Read Active Directory button.
All user information is displayed in the table. (A record must have either an Ext Number or First Name or
Last Name or Mail Address, otherwise it will not be not listed in the table.)
5.
Select the extensions you want to import. You can use the Select All button, but only records that have an extension number can be selected. If an extension number is empty, a warning message pops up.
6.
You can use the Clear All button to clear all check marks.
7.
Click the Import button. A progress bar tells you the progress of importing.
8.
If an extension already exists in the destination list, a dialog box opens.
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9.
Respond to the question in the dialog box. If you decide to overwrite the extension, other fields not in the
Active Directory are kept.
After finishing importing, a dialog box pops up to tell you how many extensions were imported successfully.
When you click OK, an error report file is opened automatically to tell you the detailed information on every extension. If some fields are invalid, the system replaces them with a default value (except for the extension number). (The report file's name is "ReportImportAD.txt". It is in the \altiserv\exe directory.)
Exporting the Extensions in a MAXCS System
1.
Go to Services > Utilities > Export Extensions.
2.
Click the Explore button and specify a name and location for the .csv file you’re about to create.
3.
Check the fields you want to export. Use the Select All and Clear All buttons to select or clear all the check boxes.
Note: You must export the extension number field.
4.
Click the Export button to save the extension configurations to a .csv file.
A progress bar shows you the progress of the export. When the export is complete, a dialog box indicates how many extensions were exported.
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AltiGen Custom Phrase Manager
Editing a .csv File
If you edit a .csv file,
•
All fields must be separated by a "," and all the records must be divided by pressing the Enter key.
•
The first line must be a pre-defined field name, such as “First Name”. If the field name doesn’t match a pre-defined field name, the field is skipped during an import operation.
•
The sequence of the columns doesn’t matter.
AltiGen Custom Phrase Manager
The AltiGen Custom Phrase Manager is a Windows-based application that makes managing custom phrases easy. It displays all custom phrases in a graphical user interface. You can add or delete a phrase by clicking a button. You also can rename an existing phrase to a meaningful name, rather than pressing digits on the telephone.
Note: The AltiGen Custom Phrase Manager requires a Client SDK license.
To use the AltiGen Custom Phrase Manager,
1.
Open the tool from the Windows Start > All Programs menu. You’ll see the login screen:
Enter the following:
•
MAXCS server address
•
Manager Extension
•
Manager Extension password.
If you want to save the password for this application, check the Always Save Password check box.
Note: The server address and the extension number will be written to the windows registry. If you choose
Always Save Password, the password will be encrypted and also saved in the registry. The tool will automatically reload the server address, manager extension number and the password from the registry when it starts next time.
2.
Click Login. The main window opens:
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Chapter 29: Tools and Applications
•
The list at the top left displays all the directories of custom phrases under your MAXCS system’s
PostOffice\phrases\ directory, such as LangCustom, LangCustom_Chinese, Tenant1Custom.
•
The list at the top right lets you select an extension through which to record or listen to a phrase.
•
The table shows all custom phrases under the selected directory, including:
•
Phrase name
•
Date and time the phrase was created or last modified
•
Phrase length
•
A column for a description of the phrase
Data can be sorted in ascending or descending order by clicking a column heading.
•
Buttons let you play, create and edit phrases.
Creating a New Phrase
To create a new phrase,
1.
Select the extension you will be using to record the phrase.
2.
Click the New button.
3.
Enter a name for the phrase.
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AltiGen Custom Phrase Manager
4.
Click Start Recording.
5.
When finished recording, press # on the phone and follow the instructions you hear. Also click OK in the dialog box when done.
Playing a Phrase
To play a phrase,
1.
Select the extension you will be using to listen to the phrase.
2.
Click the Play button. The extension will ring.
3.
Answer the ring, and a voice announces the phrase before playing it.
4.
When you are finished listening, hang up the phone and click the OK button in the AltiGen Custom Phrase
Manager.
Editing a Phrase Name or Description
To edit the name of a phrase or its description,
1.
Select the phrase you want to edit.
2.
Click the Edit button.
3.
Make your changes to the name and description. Click OK.
Deleting Phrases
To delete a phrase,
1.
Select the phrase you want to delete.
2.
Click the Delete button. A confirmation dialog box opens.
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3.
To delete the phrase, click Yes. The phrase is deleted from the directory and from the table in AltiGen
Custom Phrase Manager.
Re-recording Phrases
To re-record a phrase,
1.
Select the extension you will be using to re-record the phrase.
2.
Select the phrase, and click the Re-record button.
3.
Click the Re-Record button.
4.
When finished recording, press # on the phone and follow the instructions you hear. Also click OK in the dialog box when done.
HMCP Certification Test Tool
The HMCP Certification tool helps you measure how many channels can be simultaneously operated in an
HMCP system, based upon a threshold of CPU usage. The tool can also be used to measure DPC (Deferred
Procedure Call) latency of a Microsoft system.
For instructions on installing this tool and running the various HMCP tests, refer to the detailed steps in the
Softswitch Deployment Guide
, found in the AltiGen Knowledgebase. You will need to first install the tool, and then run it.
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C
H A P T E R
30
Quality of Service (QOS) Configuration
This section describes how to add QoS tags for MAXCS RTP packets.
Configure the MAXCS Server
1.
Open the Local Group Policy Editor on the MAXCS server: In Windows, click Start. In the Search box, enter gpedit.msc and press Enter.
2.
Open MMC: In Windows, click Start. In the Search box, enter mmc and press Enter.
3.
On the File menu, click Add/Remove Snap-in. In the dialog box, click Group Policy Object Editor and click Add.
Figure 265. The Add or Remove Snap-ins dialog box
4.
In the next dialog box, click Browse.
5.
You can click either This computer to edit the Local Group Policy object, or click Users to edit
Administrator, Non-Administrator, or per-user Local Group Policy objects.
6.
Click Ok. Click Finish.
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Chapter 30: Quality of Service (QOS) Configuration
7.
Switch back to the Local Group Policy Editor. In the left pane, expand Local Computer Policy >
Computer Configuration > Windows Settings.
8.
Right-click Policy-based QoS and select Create new policy.
Figure 266. The Local Group Policy Editor window
9.
In the new window, enter a policy name and specify a DSCP value from 0-63. Click Next.
Figure 267. Enter a name for the policy and a DSCP value
424 MAXCS Administration Manual
10. Select All applications and then click Next.
Configure the MAXCS Server
Figure 268. Specify that the policy applies to all applications
11. Select Any source IP address and Any destination IP address in this panel and click Next.
Figure 269. Specify source and destination IP addresses
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Chapter 30: Quality of Service (QOS) Configuration
12. Open MAXCS Administrator. Obtain the port range (in the Local Ports column) in the Current Resource
Statistics
window.
Figure 270. MAXCS Administrator Current Resource Statistics
13. Switch back to the QoS policy window. a.
Select UDP for the protocol. b. Select From this source port number or range and enter the port range from step 12.
426 MAXCS Administration Manual
c.
Select To any destination port.
d. Click Finish.
Configure Each MaxAgent Client System
Figure 271. Figure 7: Set QOS options
Configure Each MaxAgent Client System
Repeat the steps listed in the previous section on each MaxAgent client system: add the policy with the same steps, but use port number 30000.
Notes
If you had a Windows client machine on which you are pushing Policy Based QoS, and you noticed in a subsequent network trace that the DSCP value is 0x00, perform these steps:
1.
If you need to make DSCP values to take effect on the adapter which does not have Domain access, you need to add the following registry on the system:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\services\Tcpip\QoS
Type: REG_SZ
Name: Do not use NLA
Value: 1
Create the key "QoS" if it does not exist.
After you create the QoS registry key, reboot the computer.
2.
If step 1 does not work, create an additional registry entry:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\services\Tcpip\Parameters] "Disable-
UserTOSSetting"=dword:00000000
Reboot the server and check the behavior.
If the system belongs to a domain, log in as a domain user; logging in as the local administrator or a user will not work.
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A
P P E N D I X
A
E1-R2 and E1 ISDN PRI Installations
This section describes the configuration procedures necessary to implement E1 R2 digital signaling for
European, Pacific Rim, and other emerging markets. Please carefully follow the procedures step by step.
1.
Change the Configure Type to E1: a.
From Boards view, double-click the board to be configured to open the Board Configuration window.
b.
In the Board Configuration window, click the Board Configuration button. c.
In the next Board Configuration window, select E1 as the configure type, and click OK.
Important:
When changing from E1 to T1, then back to E1, trunk channel properties and channel group properties will be reset to default values. It is important to make sure the trunk channel properties are configured properly. Continue to follow the steps below to re-check your settings for the physical layer, data link layer and signaling layer.
2.
In the Board Configuration window, double-click the channel group.
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Appendix A: E1-R2 and E1 ISDN PRI Installations
Figure 272. Physical Layer
Consult your CO for “Frame Type,” “Line Code,” or “Zero Code Suppression.” Do not check the Sys-
tem Clock Master check box because the CO is a clock provider, and the AltiGen system is synchro-
3.
Click the Protocol button in the Channel Group Configuration dialog box to open the Protocol
Configuration window.
Figure 273. Data Link Layer
4.
Select E1 CAS as the Span Type, and select the Switch Mode according to your country in the E1 R2
Setting field, and click OK.
5.
In the Trunk Configuration window, click the Trunk Properties button to open the E1 Channel
Configuration window.
430 MAXCS Administration Manual
Figure 274. Signaling Layer
6.
In the E1 Channel Configuration dialog box, configure the following fields:
•
Set Calling Party Category to – the Calling Party Category indicates the type of calling party, (for example, operator, pay phone, priority, ordinary subscriber). Select 1, 2 or 3 (for ordinary subscribers,
the proper value.
•
Add extra digits before caller ID – consult your CO to find out if any extra digits are needed.
•
Incoming sequences – select check box and configure the sequence according to Table on page
•
In-call signaling – configure the in-call signaling value according to Table on page 432.
property.
Note: Consult your CO to find out if caller ID digits are provided in the lines.
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Signaling Values, By Country
Table 1. Signaling values, by country
Country
Chile/Nacional MFC-R2
China MFC-R2
Signaling Values
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown): DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
Set calling part category: 3
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (116)*(K)31
For 4-digit DID, set to (1116)*(K)31
For 5-digit DID, set to (11116)*(K)31
For 6-digit DID, set to (111116)*(K)31
For 7-digit DID, set to (1111116)*(K)31
For 8-digit DID, set to (11111116)*(K)31
432 MAXCS Administration Manual
Country
Colombia MFC-R2
Ecuador MFC-R2
Signaling Values
Set calling part category: 2
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
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Appendix A: E1-R2 and E1 ISDN PRI Installations
Country
Ecuador MFC-LME
Korea MFC-R2
Mexico / Teléfonos de
Mexico
Signaling Values
Set calling part category: 2
[The switch doesn't support caller ID transmission]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
Set calling part category: 1
[The switch doesn't support caller ID transmission]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
Set calling part category: 2
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (116)*(K)31
For 4-digit DID, set to (1116)*(K)31
For 5-digit DID, set to (11116)*(K)31
For 6-digit DID, set to (111116)*(K)31
For 7-digit DID, set to (1111116)*(K)31
For 8-digit DID, set to (11111116)*(K)31
434 MAXCS Administration Manual
Venezuela / Nacional
MFC-R2
Country
Panamá / Nacional MFC-
R2
Signaling Values
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
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Appendix A: E1-R2 and E1 ISDN PRI Installations
Country
China MFC-R2
Colombia MFC-R2
Signaling Values
Set calling part category: 3
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (116)*(K)31
For 4-digit DID, set to (1116)*(K)31
For 5-digit DID, set to (11116)*(K)31
For 6-digit DID, set to (111116)*(K)31
For 7-digit DID, set to (1111116)*(K)31
For 8-digit DID, set to (11111116)*(K)31
Set calling part category: 2
[Assume no caller ID provided] :
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
436 MAXCS Administration Manual
Country
Ecuador MFC-R2
Ecuador MFC-LME
Korea MFC-R2
Signaling Values
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided] :
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
Set calling part category: 2
[The switch doesn't support caller ID transmission]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
Set calling part category: 1
[The switch doesn't support caller ID transmission]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
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Appendix A: E1-R2 and E1 ISDN PRI Installations
Mexico / Teléfonos de
Mexico
Country
Panamá / Nacional MFC-
R2
Signaling Values
Set calling part category: 2
[Assume no caller ID provided] :
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)1
For 4-digit DID, set to (1113)1
For 5-digit DID, set to (11113)1
For 6-digit DID, set to (111113)1
For 7-digit DID, set to (1111113)1
For 8-digit DID, set to (11111113)1
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (116)*(K)31
For 4-digit DID, set to (1116)*(K)31
For 5-digit DID, set to (11116)*(K)31
For 6-digit DID, set to (111116)*(K)31
For 7-digit DID, set to (1111116)*(K)31
For 8-digit DID, set to (11111116)*(K)31
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits) :
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
438 MAXCS Administration Manual
Country
Venezuela / Nacional
MFC-R2
Signaling Values
Set calling part category: 1
[Assume no caller ID provided]:
Incoming sequence: DID/DNIS
In-call signaling (depend on how many DID digits):
For 3-digit DID, set to (113)6
For 4-digit DID, set to (1113)6
For 5-digit DID, set to (11113)6
For 6-digit DID, set to (111113)6
For 7-digit DID, set to (1111113)6
For 8-digit DID, set to (11111113)6
[Assume caller ID provided]:
Incoming sequence (same as above shown):
DID/DNIS * Caller ID
In-call signaling (depend on how many DID digits) :
For 3-digit DID, set to (115)*(K)36
For 4-digit DID, set to (1115)*(K)36
For 5-digit DID, set to (11115)*(K)36
For 6-digit DID, set to (111115)*(K)36
For 7-digit DID, set to (1111115)*(K)36
For 8-digit DID, set to (11111115)*(K)36
E1 ISDN PRI Installation
This section describes the configuration procedures necessary to implement E1 ISDN PRI signaling for
European, Pacific Rim, and other emerging markets. Please carefully follow the procedures step by step.
1.
Change the Configure Type to E1: a.
From Boards view, double-click the board to be configured to open the Board Configuration window. b.
In the Board Configuration window, click the Board Configuration button. c.
In the next configuration window, select E1 as the configure type and click OK.
Important:
When changing from E1 to T1, then back to E1, channel group properties will be reset to default values. It is important to make sure the channel group properties are configured properly. Follow the steps below to re-check your settings for the physical layer and data link layer.
2.
In the Board Configuration window, double-click the channel group.
MAXCS Administration Manual 439
Appendix A: E1-R2 and E1 ISDN PRI Installations
Figure 275. Physical Layer
Consult your CO for “Frame Type,” “Line Code,” or “Zero Code Suppression.” Do not check the Sys-
tem Clock Master check box because the CO is a clock provider, and our system is synchronized to the
CO. If all configurations are correct, the status should be shown as “OK,” as in Figure 275.
3.
Click the Protocol button in the Channel Group Configuration dialog box to open the Protocol
Configuration window.
4.
Select Regular ISDN PRI as the Span Type, and select the Switch Mode according to your country in the
ISDN PRI Setting field, and click OK.
Figure 276. Data Link layer
What you should select in the B Channel Maintenance Message list depends on what country you reside in
440 MAXCS Administration Manual
Protocol/B Channel Maintenance Message Settings, By Country
Table 2. Protocol & B Channel Maintenance Message Setting, by Country
Nordic
Poland
Russia
Saudi Arabia
Singapore
South Africa
Spain
Taiwan
Thailand
UK
UK, Ireland
USA
USA, Canada
Country
Argentina
Australia
Belgium
Brazil
China, HK
Czech
France
Germany
Greece
Italy
Japan
Korea
Macedonia
Mexico
Netherlands
ETSI
Protocol Supported
Austel TS014, ETSI
ETSI
ETSI
ETSI
ETSI
ETSI, VN4
ETSI
ETSI
ETSI
ETSI
ETSI
ETSI
ETSI
NTT INS1500
ETSI
ETSI
ETSI
ETSI
ETSI
ETSI
ETSI
Bellcore, ETSI
ETSI
ETSI (for DASS II/DPNSS)
ETSI, British Telecom ISDN 30
Bellcore TR 1268
AT&T TR 41449/41459
Restart
Restart
Restart
Restart
None
Restart
Restart
Restart
B Channel Maintenance
Message Setting
Restart
Restart
Restart
Restart
Restart
Restart
None
Restart
Restart
Restart
Restart
Restart
Restart
Restart
None
Restart
Restart
None
Restart and Service
Restart
MAXCS Administration Manual 441
Appendix A: E1-R2 and E1 ISDN PRI Installations
442 MAXCS Administration Manual
A
P P E N D I X
B
Required Service Parameters
This section identifies the recommended and supported parameters for T1, PRI, and E1 service and provides you with the information needed when you make your service request.
Service Parameters/Request Information for T1
To subscribe to T1 service, certain parameters are required to establish service. The information provided below identifies the recommended and supported parameters for T1 service. When ordering T1 service, provide the following service request information:
Equipment Information
•
PBX Manufacturer – AltiGen Communications, Inc.
•
CSU/DSU – ADTRAN T1 ACE (recommended) or other CSU/DSU
Technical Information for T1 with Voice
Signaling Protocol:
•
E&M Wink Start (recommended)
•
E&M Immediate Start
•
Ground Start
•
Loop Start (not recommended)
Trunk Type:
•
DID
•
2-Way DID (recommended)
•
DOD
Framing:
•
Super Frame (SF)/D4
•
Extended Super Frame (ESF) (recommended)
Line Coding:
•
Alternate Mark Inversion (AMI)
•
B8ZS (recommended)
MAXCS Administration Manual 443
Appendix B: Required Service Parameters
DNIS, Caller ID:
•
DTMF (Dual Tone Multi-Frequency)
Physical Termination:
•
RJ-48X or RJ-48C
Wire:
•
4 wires
800 Service:
•
You decide
Termination Impedance:
•
100 ohms
Type of Registered Services Provided
•
BN 1.544 Mbps SF without power
•
DN 1.544 Mbps SF B8ZS without power
•
1KN 1.544 Mbps ANSI ESF without power
•
1SN 1.544 Mbps ANSI ESF, B8ZS without power (recommended)
Service Order Code
SOC 6.0P AS.2
T1 Channel Assignment
•
Trunk Type – In, Out, or 2-Way (recommended)
•
Channels Assigned – 24 (Enter partial channels if you wish to subscribe to both voice and data service.)
•
Hunting – Most Idle, Least Idle, Ascend, Descend
•
DNIS Digits/Signal – 3/DTMF (can be 3 to 10 digits)
•
Caller ID Signal – DTMF, if available (Not every service provider provides Caller ID over T1 lines.)
CSU/DSU Requirements
The CSU (channel service unit) is a device used to connect a digital trunk line coming in from the phone company to the PBX. A CSU can terminate signals, repeat signals and respond to loopback commands sent from the central office.
Service Parameters/Request Information for PRI
To subscribe to PRI service, certain parameters are required to establish service. The information provided below identifies the recommended and supported parameters for PRI service. When ordering PRI service, provide the following service request information:
Equipment Information
•
PBX Manufacturer – AltiGen Communications, Inc.
•
CSU/DSU – ADTRAN T1 ACE (recommended) or other CSU/DSU
444 MAXCS Administration Manual
Technical Information for PRI with Voice
Switch Type:
•
5ESS (recommended)
•
DMS (recommended)
•
NI-2 (recommended)
•
4ESS
Framing:
•
Super Frame (SF)/D4
•
Extended Super Frame (ESF) (recommended)
Line Coding:
•
Alternate Mark Inversion (AMI)
•
B8ZS (recommended)
Physical Termination:
•
RJ-48X or RJ-48C
Wire:
•
4 wires
PRI Channel Assignment
D Channels Assignment – 24th channel (channel ID 23)
Note: MAXCS can configure any channel in a PRI span to be the D channel. The default setting is the last channel. Every span should select a D channel within the span. Shared D channel (NFAS) or back up D channel is not supported.
Hunting – Most Idle, Least Idle, Ascend, Descend
DNIS Digits – can be 3 to 10 digits
CSU/DSU Requirements
The CSU (channel service unit) is a device used to connect a digital trunk line coming in from the phone company to the PBX. A CSU can terminate signals, repeat signals, and respond to loopback commands sent from the central office.
Service Parameters/Request Information for E1
To subscribe to E1 service, certain parameters are required to establish service. The information provided below identifies the recommended and supported parameters for E1 service. When ordering E1 service, provide the following service request information:
Equipment Information
•
PBX Manufacturer – AltiGen Communications, Inc.
•
CSU/DSU – ADTRAN T1 ACE (recommended) or other CSU/DSU
MAXCS Administration Manual 445
Appendix B: Required Service Parameters
Technical Information for E1 with Voice
Switch Type:
•
Austel TS014
•
ETSI NET5 (recommended)
•
NT DMS-100
Framing:
•
No CRC (recommended)
•
CRC4
Line Coding:
•
Alternate Mark Inversion (AMI)
•
HDB3 (recommended)
Physical Termination:
•
RJ-48X or RJ-48C
Wire:
•
4 wires
E1 Channel Assignment
•
Data Channels Assignment – 1st channel (channel ID 0)
•
Channels Assignment – 17th channel (channel ID 16)
•
Hunting – Most Idle, Least Idle, Ascend, Descend
•
DNIS Digits – can be 3 to 10 digits
CSU/DSU Requirements
The CSU (channel service unit) is a device used to connect a digital trunk line coming in from the phone company to the PBX. A CSU can terminate signals, repeat signals, and respond to loopback commands sent from the central office.
446 MAXCS Administration Manual
A
P P E N D I X
C
Network Ports
If MAXCS is installed behind a firewall/NAT router, you must open TCP and UDP ports according to this table:
For external VoIP connection through a firewall UDP
Remote IP phone/IPTalk to phone service
TCP
10032
10064
5061
10060
Remote AltiGen IP phone firmware download (TFTP)
Extension global appearance 10066
69
VoIP RTP Port (Voice Stream) for SIP
SIP Tie Trunk from other AltiGen systems
SIP Trunking Service from carrier
From X to Y
(See note below)
10060
5060
Polycom phone service
80
443
When adding additional combo licenses, the system will also increase the RTP ports it uses and will use these new ports. If these additional ports are not added to the firewall, then calls will not have audio.
Note: An easy way to find out the RTP/TCP port range(s) for SIP is to look in the MaxAdministrator View >
Current Resource Statistics window. All the ports are listed in the Local Ports column.
When MAXCS or Softswitch is running on a non-Windows 2008/2012 system, BasePort = 49152.
When MAXCS or Softswitch is running on a Windows 2008/2012 system, BasePort = 49664 (This is because Windows 2008/2012 has some system services use ports in the 49152 range).
For a single chassis system:
X = BasePort
Y = BasePort + Total IP codec channels x 2
For a multi-chassis system, you need to enter multiple ranges:
Gateway ID = 0
MAXCS Administration Manual 447
Appendix C: Network Ports
X0 = BasePort
Y0 = BasePort + Total IP codec channels in GW0 x 2
Gateway ID = 1
X1 = BasePort + 512 x 1
Y1 = X1 + Total IP codec channels in GW1 x 2
Gateway ID = 2
X2 = BasePort + 512 x 2
Y2 = X2 + Total IP codec channels in GW2 x 2
Gateway ID=n
X(n)=BasePort + 512 x n
Y(n)=X(n) + Total IP codec channels in GW(n) x 2
To connect the following applications through a firewall
AltiConsole
MaxCommunicator/MaxAgent/IPTalk VM service for
MaxCommunicator/MaxAgent
MaxCommunicator/MaxAgent MeetMe Conference
MaxSupervisor
MaxMobile Communicator
Client Applications Auto Update
CDR Search
Remote MAXCS Administrator
VRManager
(VRManager may not work behind NAT)
MaxInSight
Network Assessment Tool
10025
TCP
10025
10026
10028
10040
10025
10027
10028
10029
10050
10080
10081
10050
10025
10068
10040
10029
10010
UDP
448 MAXCS Administration Manual
Remote IP Phones Behind NAT
MAXCS connects the following application through a firewall
External CDR Logger Service 10027
TCP UDP
MAXCS 8.0 uses internal network port 10072 to work with the client applications. Other applications on the users' system should not use this port. Since this is for internal use, no firewall setting should be configured for this port.
Remote IP Phones Behind NAT
For remote IP phones behind NAT, you don’t need to do any configuration. However, if the remote firewall/NAT router blocks outgoing traffic, then you will need to open the following ports on the remote firewall/NAT router:
•
UDP 10060
•
UDP 30,000~31,000
•
TCP 10064
MAXCS Administration Manual 449
Appendix C: Network Ports
450 MAXCS Administration Manual
A
P P E N D I X
D
Technical Support & Product Repair Services
This section describes:
•
AltiGen technical support policy and procedures
•
Product repair
•
Technical training for administrators
Technical Support Eligibility
Eligibility: AltiGen provides technical support to Authorized AltiGen Partners and distributors only.
End user customers, please contact your Authorized AltiGen Partner for technical support.
How To Reach AltiGen Technical Support
Authorized AltiGen Partners and distributors may contact AltiGen technical support by the following methods:
•
You may request technical support on AltiGen’s Partner web site, at https://partner.altigen.com. Open a case on this site, and a Technical Support representative will respond within one business day.
•
Call 888-ALTIGEN, option 5, or 408-597-9000, option 5, and follow the prompts. Your call will be answered by one of AltiGen’s Technical Support Representatives or routed to the Technical Support
Message Center if no one is available to answer your call.
Technical support hours are 5:00 a.m. to 5:00 p.m., PT, Monday through Friday, except holidays.
If all representatives are busy, your call will be returned in the order it was received, within four hours under normal circumstances. Outside AltiGen business hours, only urgent calls will be returned on the same day
(within one hour). Non-urgent calls will be returned on the next business day.
Please be ready to supply the following information:
•
Partner ID
•
AltiGen Certified Engineer ID
•
Product serial number
•
AltiWare or MAXCS version number
•
Number and types of boards in the system
•
Server model
•
The telephone number where you can be reached
MAXCS Administration Manual 451
Appendix D: Technical Support and Product Repair Services
•
A brief description of the problem and the procedure to reproduce the problem
Having this information ready will help us to better assist you.
End users who have problems unresolved by their AltiGen Authorized Partner, and Partners who have problems unresolved by AltiGen Technical Support may send an e-mail to AltiGen’s CEO at [email protected].
Product Repair
You may send defective AltiGen-manufactured hardware products (in or out of warranty) to our factory for prompt authorized repairs. For information on AltiGen repair services and return policies and the AltiGen warranty, visit the AltiGen Partner portal at https://partner.altigen.com.
Technical Training for Administrators
AltiTraining, LLC, has created comprehensive 3- and 4-day hands-on training courses to teach AltiGen system administrators everything from the basic skills of extension configuration to troubleshooting and multiple location implementation of Voice over Internet Protocol (VoIP).
The intensive courses were developed under the guidance of the AltiGen corporate office with the help of
Partners, installers, and customers. AltiTraining’s comprehensive curriculum is based on the same format
AltiGen uses to train their engineers and Partners. Experienced telecommunications professionals teach
AltiTraining classes and they bring a wealth of real-life experience to every course. AltiTraining classes consistently are rated as one of the most valuable and relevant that our students have ever attended!
Who Should Attend?
Anyone responsible for the day-to-day administration of an AltiGen IP-PBX telephone system or anyone who would like to learn about the system features and functionality will benefit from this hands-on training.
What Do the Courses Cover?
•
Phone line options (analog lines vs. digital T1/PRI with DID).
•
How to install and upgrade hardware, software, licenses and wiring options.
•
Server design (backplane, OS, RAID, RPS, and so on.), configuration and growth planning.
•
Day-to-day administration with MaxAdministrator. Learn to build extensions, huntgroups and workgroups to set up call handling and routing features, and system configuration options.
•
Utility and security programs to simplify and provide security/fraud insight and set up routine system backups for disaster recovery.
•
Call detail reporting (CDR) and real-time monitoring (MaxSupervisor).
•
MaxCommunicator, MaxAgent, MaxSupervisor, and AltiConsole client applications.
•
Voice over IP (VoIP) network requirements and implementation for branches or remote workers.
•
Unified messaging, utilizing Microsoft CRM or Outlook
•
System troubleshooting, covering common problems/scenarios and basic troubleshooting techniques.
•
New product developments and future upgrades.
452 MAXCS Administration Manual
A
P P E N D I X
E
Troubleshooting
This section describes some common problems you may encounter, and provides steps you can take to try to resolve them.
Polycom Phone Cannot Register
If a Polycom IP phone cannot register, try the following:
1.
Check LAN connectivity
2.
Check that the firewalls have ports 80 and 443 unblocked
3.
Reset the Polycom phone to the manufacturer’s default and then reboot it. For instructions, see the
MAXCS Polycom Configuration Guide
.
Poor Voice Quality
When voice quality is poor, try the following:
1.
Perform a Loop-Back Test. Call yourself by dialing out and dialing back into yourself. If you don’t have any problems performing this test, the problem is most likely in the network or at the remote site.
2.
Check Traffic Between MAXCS IP Stations. Open the Current Resource Statistics window (on the View menu) and the IP Cumulative Traffic Statistics window (on the Report menu) in MaxAdministrator to view network traffic.
3.
Check the RTP and RTCP Settings. RTP/RTCP stands for Real-Time Transport (Control) Protocol, a transport protocol for real-time applications used to transport packetized voice packets over the IP network. Make sure UDP port numbers (49152 + n*512) ~ (49152 + n*512+p*2), where “n” is the gateway
ID and “p” is the number of IP resource channels, are not assigned to other applications.
Note: You can find this range displayed in the Current Resource Statistics window in the Local Ports column.
4.
Check Network Configurations. Follow all network configuration guidelines provided under “Network
are configured properly.
MAXCS Administration Manual 453
Appendix E: Troubleshooting
Cannot Make a Connection
If a connection cannot be made, check the following:
1.
Check network connectivity using “ping.”
2.
3.
Check the IP address of the destination system.
4.
Check the RTP and RTCP settings. Make sure UDP port numbers 49152-49199 are not assigned to other applications. RTP/RTCP stands for Real-Time Transport (Control) Protocol, a transport protocol for realtime applications used to transport packetized voice packets over the IP network.
5.
6.
Check if Called Extension is a Workgroup or has Multiple Call Waiting Enabled. When the called party is a workgroup pilot number or has Multiple Call Waiting enabled, the caller is placed on hold and hears ringback or music.
IP Resource Does Not Appear in Current Resource Statistics
When an IP resource doesn’t appear in the Current Resource Statistics window, there are two possible causes:
•
Device Driver is Not Running. Check the device driver. Make sure it’s installed and working properly.
•
Triton VoIP Board is Not Installed Properly. Refer to the Quick Installation Guide for details on proper installation of the Triton VoIP board.
454 MAXCS Administration Manual
Index
Symbols
#12, enabling, for language setting
#17, Polycom Station Log-in
#27 to relocate global extension
Numerics
10 digit dialing area codes
A access code
access, system
account code blocking display
forcing
activity configuration
adding a huntgroup
adding a workgroup
admin defined #
admins, how many connected to system
advanced queue management
menu selection
queue overflow
after hours scheduling
agent check box
agents auto logout
allow call redirect/priority change
alternate mark inversion (AMI)
alternate server behind NAT
setting
switching to
AltiGen board test tool
AltiGen IP phone configuration
AltiGen services stop & start
Alti-Mobile Extension limitations
AM schedule
AMI line coding
announcement time stamp
answer options
answering huntgroup call handling
workgroup call handling
application extension
definition & uses
failover plan
setup
application extension configuration
application failover plan
Apply To, multiple extensions
area code, on trunk
area codes system home
assigning client licenses
attributes setting trunk
trunk
audio peripheral configuration
audio peripheral options for huntgroups
auto attendant adding
collecting digits
configuring
editing
making assignments
menu items, configuring
prompts, phrase management
push URL/Web page
recording custom phrases
setting call priority
setting call SKLR
auto logout agents
auto record personal extension calls
auto-discovery of server IP address configuring
disabling
two servers in network
B
B8ZS (Binary 8 Zero Substitution)
back up system data
backing up files
Backup & Restore Utility
bandwidth
and public pipe
WAN
basic queuing control
binary 8 zero substitution (B8Z8)
BLF programmable key
Block Caller Name and Number from view
blocking account code display
blocking all outgoing calls
blocking caller ID
blocking calls
blocking unauthorized SIP Invite requests
board
MAX1000/2000, configuring
mobile extension, configuring
SIPSP, configuring
Triton Analog Station, configuring
Triton Analog Trunk LS/GS &
LS, configuring
Triton MeetMe
Triton Resource, configuring
Triton T1/E1, configuring
Triton T1/E1, setting up channels
Triton VoIP, configuring
board configuration
Boards View window
boards, supported
bridge conference
business hours
24-hour business hour setup
business hours profile caller ID routing
DNIS routing
business hours, setting up
busy call handling
,
huntgroups
C call accounting report
call blocking, outgoing
Call Center menu
call control
call handling
,
for workgroups
huntgroup
incoming
Call Log View window
call parking
call priority caller ID routing
DNIS routing
setting
call record programmable key
MAXCS Administration Manual 455
call recording configuring system-wide
extension based recording
file name description
multiple gateways
personal options
remote shared directory
requirements
trunk based recording
call recording configuration
call reports
call reports, external
call restrictions
extension
system
call routing
call screening enabling
with ONA
call SKLR setting
call waiting distinctive
distinctive tones
multiple
options
Callback from Queue
callback interview
callback number
,
caller ID
collecting
for E911 calls
transmitted
caller ID routing
business hours profile
call priority
holiday profile
caller ID verification
calling numbers, PRI
calls, blocking all outgoing
card logical ID
CDR, setting up
Centrex transfer
Centrex, enable Centrex transfer
changing password
changing scope
channel
channel group info
channel information
channel mapping list
channel number
channel service unit, installing
client licenses
CO switch, and terminal device
code access
area
codec G.722
codec negotiation
codec parameters
codec priority
codec profile assigning to IP addresses
inter-gateway, setting
setting
codec silence suppression
collecting caller ID and DID digits
collecting digits, in auto attendant
collecting trace
conference bridge option
conference calls, types
Configuration Reader tool
confirm callback number
,
connection difficulty
copy from button, IP phone configuration
country, for system
cross talk, test tool
CSU installation
CT-Bus clock master, and T1/E1
Clocking
CT-Bus test tool
cumulative IP traffic statistics
Current Resource Statistics window
missing IP resource
current traffic statistics refresh interval
custom application and message notification
,
custom phrase manager
D data backup
restore
dedicated mobile trunk
default password for MaxAdmin
default routes, outcall routing
device driver
DHCP option 120
diagnosing tools
Diagnostic menu
dialed digit translator
dialing en-bloc
overlap
dialing 9 twice, preventing
dialing delay resolving
T1 signaling
dialing pattern tips
dialing pattern tips, out call routing
dialing patterns
dialing scheme trunk
DID collecting
DID number extension
display workgroup status
IP phone
distinctive call waiting
distinctive call waiting tones
distinctive ringing
distribution lists, configuring
distribution lists, creating
DNIS and language setting
DNIS routing
business hours profile
call priority
holiday profile
do not disturb, setting
domain, VoIP adding servers
creating multi-site
extension scope
managing users
master
rejoining a server to
relocating extension
relocating extension using #27
synchronizing manually
system ID and
DTMF delivery
E
E1 channel assignment
subscribing to service
E1 ISDN PRI installation
E1-R2 CAS installation
e911 CID for extensions
e911 support
location IDs
on AltiGen phones
name
e-mail messaging options
e-mail services
456
MAXCS Administration Manual
e-mail, setting notification
emergency dial notification
emergency numbers
en-bloc dialing
Enterprise Manager
changing the password
default password
fixed IP address
how to log in
login failed
resetting password
user interface
ESF (Extended Superframe Format)
Exchange creating new mailbox user
integration configuration
debugging
options, setting
requirements
troubleshooting tips
synchronization, testing for
Exchange server synchronizing voice mail with
exporting extensions
Extended Superframe Format
(ESF)
extension assigning groups to
calling options
changing location & type
DID number
e-mail name
exporting to csv file
general settings
importing from Active Directory
importing to csv file
incoming call handling
information only mailbox
length
mail forwarding
mail management
MeetMe Conference
message notification
monitor list
monitoring
outgoing call restrictions
physical extension
relocating in domain
relocating in domain using #27
security checker
send notification
setting phone display options
setting SMTP/POP3
setting Triton analog line properties
speed dial
types defined
virtual
extension activity
extension based recording
extension configuration
extension dialed digit translator
extension scope changing
Extension Security Checker
Extension View window
external logging
F failover when network is down
Fax-over-IP (FoIP) and out call routing
codec setting
configuration
enable fax trunk routing
,
for IP extensions
hunt groups
feature code programmable key
files backup
firewall, configuration
firewall, opening ports
firewall/NAT router configuring port forwarding
first digit
first digit translator
flash key (Alti-IP 600)
forced account code
forward from group greeting
forwarding numbers, specifying
G
G.722 codec
global extension rerouting
global least cost routing, configuring
greeting prompts
group setup
group greeting, press "0" for forward
group paging
groups assigning to extension
GTE
H hackers, detecting
hardware problems
hardware status
headset key
Help menu
HMCP Certification Test Tool
holiday routing rules
holiday profile caller ID routing
DNIS routing
home area code
hop off enabling
hunt group converting to workgroup
huntgroup adding a
answer handling
business hours
busy call handling
call handling
configuration
configuring basic attributes
mail management
queue management
setting hours
setting up membership
setup
single call handling
huntgroup hours
I
ID, server changing length
impedance match, performing
importing extensions
in call routing
in call routing rules
incall routing
Web IP calls
incoming call handling
information only mailbox
installation
E1 ISDN PRI
E1 R2 CAS
MAXCS, preparation
installing
MaxAdmin
MAXCS Administration Manual 457
MAXCS
installing a Channel Service Unit
(CSU)
integrating Exchange Server
inter-call delay, setting
intercom, enabling
Intranet
intranet pipe
configuring
IP addresses undefined
IP Cumulative Traffic Statistics window
IP device range
IP dialing table, configuring
IP extensions
IP network defining address range
IP phone display workgroup status
IP 805 configuration
registration password
time display
IP resource, missing from Current
Resource Statistics window
IP Trunk Access
ISDN PRI switch mode, setting
ISDN setting
TEI
ISP
J
Jam Bit 8
jitter average statistic
jitter buffer
jitter buffer range
L language
DNIS routing to
enabling #12
rules MAXCS follows
setting in extension config
languages configuring extension
enabling in AA
other, configuring
LCR, configuring
least cost routing, configuring
License menu
licenses (table)
licenses, clients
limitations
Alti-Mobile Extension
line loss, acceptable range
line park
line park, configuration
live call handling, enabling
local network defining
location format for trunks
log file network log
log file, security alert
logging outbound workgroup calls
login
login failed, Enterprise Manager
logout
M mail forwarding setting extension for
mail management for extensions
for workgroups
mailbox
information only
size
,
mailbox capacities
main menu
main number
making a connection difficulty with
Manager Extension
MAX channel group info
MaxAdmin installing
MaxAdmin & Extension Security
Checker
MaxAdmin default password
MaxAdmins, disconnecting from system
MaxCS behind NAT configuring
forwarding ports
MaxCS Private Cloud feature differences
MaxMobile, configuring
,
MDMF (Multiple Data Message
Format)
measuring Rx level of trunk channel
media path, about
MeetMe Conference appointing an admin
auto-mute feature
configuring
e-mail template
overview
menu selection advanced queue management
menu, MaxAdmin main
message length
maximum number of
notification
schedule
notification for workgroup
notification options, setting
notification timing, setting
notification type
playback
retention
message notification setting
message playback options
messages managing
recording options
setting e-mail options
setting notification retries
messaging, managing
Microsoft Exchange integration
Microsoft SQL
mobile extension and MaxMobile Communicator
limitations
overview
press any key to answer call
voice mail
mobile trunk shared or dedicated
Monitor Available list
monitor list
monitoring extensions, set up
multilingual system auto attendant
configuring
configuring #12 for extension
configuring DNIS routing
configuring extension
language used
multilink
multiple call waiting, enabling
multiple data message format
(MDMF)
music files converting
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music on hold
MVIP cable, test tool
N
N/A programmable key
NAT support
MaxCS behind, configuring
network defining
IP, defining address range
quality of service
Network Address Translation
(NAT)
network guidelines for VoIP
,
network log file
network management
network ports used by MAXCS
no answer handling
non-workday handling
notification message
schedule
workgroup messages
notification type and timing
,
numbering plan
ISDN, configuring
O
ONA call screening
configuring
enabling
ring duration
one number access configuring
enabling
one-way connection, test tool
online help
Operator Extension
operator group
option 120, DHCP
out call routing configuration
configuring
default routes
dialing delay
dialing pattern tips
dialing patterns
example configuration
overview
route definitions
out of sync
outgoing calls, blocking
outgoing calls, blocking all
overhead paging
overlap dialing
P pager notification
paging group paging
overhead paging
trunk configuration
paging groups
password changing
extension
IP phone registration
PBX menu
personal call recording options
phone display options
phone number trunk
phrase management for auto attendant
phrase manager, custom
physical extension
ping, to check network connectivity
pipe intranet
intranet, configuring
public
public, configuring
playback message
playing music from a file
PM schedule
Polycom phones
#17 Station Log-in
Advanced Features license
Configuration Guide
default local admin password
DTMF Delivery setting
Global extension relocation
user directories
video codec
poor voice quality
POP3 service
port number
ports, network
Postmaster Extension
prefix system prohibited
toll call
press "0" option
press any key to answer call
PRI channel assignment
subscribing to service
PRI calling numbers
PRI ISDN numbering plan
priority promotion
priority queuing
private networks
product repair
prompts converting
greeting and update
multiple languages, storing
PSTN failover
public pipe
configuring
push URL/Web page
Q quality of service (QoS)
queue announcement
queue call priority score
queue management advanced
for huntgroups
for workgroups
queue overflow
queue overflow routing
queue phrase options
queuing control basic
quit queue
R
Read Config tool
Realtime Transport Control Protocol (RTP/RTCP) definition
recorded announcements, configuration
recording auto attendant phrases
,
configuring call
configuring on trunk
file description
messages
multiple gateways
personal options
remote shared directory
requirements
recording options for workgroups
Recording Seat license
Recording Session license
recording tone
redirect IP phones when server down
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refresh enterprise configuration
refresh interval, current traffic statistics
rejoining a server to VoIP domain
Release SIP Tie-Link Trunk option
relocating domain extension
relocating domain extension using #27
remote IP phones behind NAT
remote locations
repair, product
replicate from domain
report cumulative IP traffic statistics
system summary
Report menu
reports, call logs
reports, system
rerouting outgoing calls
reset board button
Reset button, Extension View
Reset button, Trunk View window
resetting cumulative statistics
Resource board
Restoring files
restricting tie trunk calls
restrictions call
outgoing call
ring all available members
,
ring average longest idle member
ring back
ring fewest answered calls
ring first available member
,
ring longest idle member
ring shortest average talk time
ringing distinctive, enabling
RNA Agent Auto Logout
RNA Agent Not Ready
route access
route access code vs trunk access code
route definitions out call routing
router
routing by caller ID
by caller ID & DNIS
by DNIS
incoming calls
routing rules, in call
RTP & RTCP settings
RTP packet length
RTP/RTCP definition
Rx level
S scheduling backup
scope of extension changing
VoIP domain
seat-based licenses
secure RTP
security alert log file
security, detecting hackers
send notification
server down redirect IP phones
server ID length
changing
server IP address, auto-discovery
server IP address, in IP dialing table
service parameters
subscribing to
service level calculations options button
service level for workgroups
service level threshold
services
AltiGen, stop & start
SMTP/POP3
Services menu
setting 10 digit dialing area codes
setting huntgroup hours
setting trunk attributes
setting up extensions
groups
huntgroup mail management
huntgroup membership
huntgroups
workgroups
SF (Superframe Format)
shared mobile trunk, setting
signal channel, about
signaling protocol
T1
single call handling
single call waiting
SIP Early Media
SIP tie trunk properties
SIP transport options
SIP transport, ext assignment vs codec profile
SIP Trunk license checking
SIP Trunk properties
SIP Trunk TLS support
SIP trusted device list
skill-based routing
SMTP service
SMTP/POP3 setting for extension
speed dial station
speed dialing configuration
SQL
starting AltiGen services
static noise, test tool
station speed dialing
Status bar information
Stop Switching Service
stop/start
MAXCS services
stopping AltiGen services
subscribing to service
Superframe Format (SF)
switchover to alternate server
synchronizing domain servers manually
system business hours
call park
call restrictions
country relevant
distinctive ringing
extension length
home area code
ID
main number
messaging
numbering plan
summary report
work days configuration
system clock master
system data backup
restore
system hang, test tool
System menu
system reports
system requirements
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MAXCS Administration Manual
T
T1 channel assignment
subscribing to service
T1 signaling dialing delay
protocol
T1/E1 clock
T1/PRI troubleshooting
TCP fragmentation
TCP ports
Maxcs behind NAT
technical support
technical training for system admins
TEI (terminal endpoint identifier)
terminal endpoint identifier (TEI)
test tools
testing AltiGen boards
text tag, collecting digits
TFTP server
tie trunks
time display
TLS support
toll call prefixes
tools, AltiGen
Trace Collector tool
trace, collecting
traffic, VoIP statistics
training, technical, for system admins
transmitted caller ID
Triton Analog Station Line Properties dialog box
Triton analog trunk GS/LS properties, setting
Triton Resource board
Triton T1/E1 trunk properties, setting
troubleshooting cannot make connection
checking network configuration
checking traffic
loop-back test
poor voice quality
troubleshooting T1/PRI
trunk access code
attributes
Centrex
configuring recording
dialing scheme
direction of transmission
incoming call routing
location format
phone number
trunk access
trunk access code vs route access code
trunk based recording
trunk channel, measuring Rx level
trunk configuration
Trunk Monitor Enable option
trunk properties
SIP tie, setting
SIP trunk, setting
Triton analog trunk GS/LS, setting
Triton T1/E1 trunk, setting
Trunk View window
trunks paging
setting attributes
Triton T1/E1 trunk properties, setting
unavailable
using Apply To button
trunks, tie
U
UDP
UDP ports
MaxCS behind NAT
unanswered calls
handling
unanswered calls, handling
uninstalling MAXCS
unusual voice mail activity notification
update prompts
Use
user defined #
users, managing VoIP domain
V
View menu
View windows
virtual extension
voice quality
Voice File Converter
voice mail configuration
distribution lists
messaging
mobile extension
setting notification
synchronizing with Exchange server
voice mail activity notification, unusual
voice mail activity, unusual setting parameters for notification
voice mail playing order
voice quality and WAN bandwidth
voicemail playing order
VoIP codec profiles
VoIP bandwidth
requirements
VoIP channel usage
VoIP configuration
VoIP domain adding servers
extension scope
managing users
master
multi-site, creating
rejoining a server to
relocating extension
relocating extension using #27
synchronizing manually
system ID and
VoIP menu
VoIP network guidelines
VoIP network management
W
WAN bandwidth
router configuration
WAN router, configuring
web IP calls
configuring incall routing
windows view (boards, trunks, etc.)
work days, setting up
Work/Hunt Group Converter
Work/Hunt Group Converter tool
workgroup adding a
answer handling
business hours
configuration
configuring basic attributes
converting to hunt group
incoming call handling
logging outbound calls
mail management
message notification
queue management
MAXCS Administration Manual 461
recording options
setup
single call handling
status display
workgroup queue agent pick up call
Workgroup View window
wrap-up time, setting
Z zero code suppression
Bell
GTE
Jam Bit 8
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