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Verbatim Store 'n' Play VM-205 Unit installation
Asterisk Appliance™ 50
(AA50)
Administrator Manual
601-00005
Rev. C
Digium, Inc.
445 Jan Davis Drive
Huntsville, AL 35806
United States
Main Number: 1.256.428.6000
Tech Support: 1.256.428.6161
U.S. Toll Free: 1.877.344.4861
Sales: 1.256.428.6262
www.digium.com
www.asterisk.org
www.asterisknow.org
© Digium, Inc. 2009
All rights reserved.
No part of this publication may be copied, distributed, transmitted, transcribed, stored in a
retrieval system, or translated into any human or computer language without the prior written
permission of Digium, Inc.
Digium, Inc. has made every effort to ensure that the instructions contained in this document
are adequate and error free. The manufacturer will, if necessary, explain issues which may
not be covered by this documentation. The manufacturer’s liability for any errors in the
document is limited to the correction of errors and the aforementioned advisory services.
This document has been prepared for use by professional and properly trained personnel,
and the customer assumes full responsibility when using it.
Adobe and Acrobat are registered trademarks, and Acrobat Reader is a trademark of Adobe
Systems Incorporated.
Asterisk and Digium are registered trademarks and Asterisk Business Edition, AsteriskNOW,
AsteriskGUI, and Asterisk Appliance 50 are trademarks of Digium, Inc.
Any other trademarks mentioned in the document are the property of their respective owners.
Digium, Inc.
Page 2
Safety Certification and Agency Approvals
Safety:
US/CSA 60950
IEC 60950
EN 60950
AS/NZ 60950
Other:
CE Mark (European Union)
2002/95/EC Restrictions on Hazardous Substances (RoHS), 2005/747/EC
lead free exemption (Annex C)
Telecom:
FCC Part 68, TIA-968
TBR-21 1998
Industry Canada IC-CS-03
AS-ACIF S002-2005
AS-ACIF S003-2005
EMC:
FCC Part 15 Class A
EN55022/CISPR22 Class A
EN55025
IEC 61000
CNS13438
VCCI V-32005.04
Digium, Inc.
Page 3
Federal Communications Commission Part 68 (USA)
This equipment complies with Part 68 of the FCC rules and the
requirements adopted by the ACTA. On the back of the Asterisk
Appliance 50 enclosure is a label that contains, among other information,
a product identifier in the format US:AAAEQ##TXXXX. If requested,
this number must be provided to the telephone company.
A plug and jack used to connect this equipment to the premises wiring
and telephone network must comply with the applicable FCC Part 68
rules and requirements adopted by the ACTA.
If the Asterisk Appliance 50 causes harm to the telephone network, the
telephone company may notify you in advance that temporary
discontinuance of service may be required. But if advance notice is not
practical, the telephone company will notify you as soon as possible.
Also, you will be advised of your right to file a complaint with the FCC if
you believe it is necessary.
The telephone company may make changes in its facilities, equipment,
operations or procedures that could affect the operation of the equipment.
If this happens, the telephone company will provide advance notice in
order for you to make necessary modifications to maintain uninterrupted
service.
If you experience problems with the Asterisk Appliance 50, contact
Digium, Inc. (+1.256.428.6161) for repair and/or warranty information. If
the equipment is causing harm to the telephone network, the telephone
company may request that you disconnect the equipment until the
problem is resolved.
Digium, Inc.
Page 4
FCC Part 15
This device complies with part 15 of FCC rules. Operation is subject to
the following two conditions: (1) This device may not cause harmful
interference, and (2) This device must accept any interference received,
including interference that may cause undesired operation.
Digium, Inc.
Page 5
Introduction to Asterisk Appliance 50 Documentation
This manual contains product information for the Asterisk Appliance 50.
Be sure to refer to any supplementary documents or release notes that
were shipped with your equipment. The manual is organized in the
following manner:
Chapter/
Appendix
Title
Description
1
Overview
Identifies the features of your unit.
2
Unit Installation
Provides instructions for installing the unit.
3
Asterisk
Configuration
Provides instructions on how to configure the
Embedded Asterisk Business Edition through the use
of the AsteriskGUI.
4
Troubleshooting
Explains resolutions to common problems and
frequently asked questions pertaining to the unit.
A
Pin Assignments
Lists the connectors and pin assignments.
B
Specifications
Details unit specifications.
C
License Agreement
Digium End-User Purchase and License Agreement
D
Glossary and
Acronyms
Defines terms related to this product.
Digium, Inc.
Page 6
Symbol Definitions
Caution statements indicate a condition where damage to the unit or
its configuration could occur if operational procedures are not
followed. To reduce the risk of damage or injury, follow all steps or
procedures as instructed.
The ESD symbol indicates electrostatic sensitive devices. Observe
precautions for handling devices. Wear a properly grounded
electrostatic discharge (ESD) wrist strap while handling the device.
The Electrical Hazard Symbol indicates a possibility of electrical
shock when operating this unit in certain situations. To reduce the
risk of damage or injury, follow all steps or procedures as
instructed.
Digium, Inc.
Page 7
Important Safety Instructions
Servicing.
Do not attempt to service this unit unless specifically instructed to do so. Do not attempt to remove the unit from your equipment while
power is present. Refer servicing to qualified service personnel.
Water and Moisture. Do not spill liquids on this unit. Do not operate this equipment in a wet environment.
Heat. Do not operate or store this product near heat sources such as
radiators, air ducts, areas subject to direct, intense sunlight, or other
products that produce heat.
Warning.
Do not place anything (including paper) on top of the Asterisk
Appliance 50. To allow proper cooling, these units must not be
stacked.
Caution. To reduce the risk of fire, use only No. 26 AWG or larger
telecommunication wiring for network connections.
Static Electricity.
To reduce the risk of damaging the unit or your equipment, do not
attempt to open the enclosure or gain access to areas where you are
not instructed to do so. Refer servicing to qualified service personnel.
Emergency 911
The Asterisk Appliance 50 is capable of forwarding arbitrary caller
id strings to VoIP service providers, which in multi-office setups
could simply be other Asterisk Appliance 50s. Customers of Internet
Telephony Service providers to which 911 or Emergency calls are
placed should ensure their provider properly forwards the customer's
accessible PSTN phone number to the emergency call handling
center.
Save these instructions for future reference.
Digium, Inc.
Page 8
TABLE OF CONTENTS
Introduction to Asterisk Appliance 50 Documentation . . . . . . . . . . .6
Chapter 1
Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
Features: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
Chapter 2
Unit Installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Unpacking the Unit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Inspecting Your Shipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Identifying Communication Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Understanding the LEDs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .19
Using the Configuration Reset Switch . . . . . . . . . . . . . . . . . . . . . . 23
Installing the Asterisk Appliance 50 . . . . . . . . . . . . . . . . . . . . . . . . 24
Mounting the Asterisk Appliance 50 . . . . . . . . . . . . . . . . . . . . . . . . 27
Chapter 3
Telephone System Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Log On to the Asterisk Appliance 50 . . . . . . . . . . . . . . . . . . . . . . . . 31
The Asterisk Appliance 50 Interface . . . . . . . . . . . . . . . . . . . . . . . . 32
Analog Hardware Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Trunk Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .40
Analog Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Outgoing Calling Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
Dial Plans . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Digium, Inc.
Page 9
Table Of Contents
User Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Ring Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .68
Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
Call Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .72
Agent Login Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Voice Menus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
Record a Voice Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
Time Intervals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Incoming Calling Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Paging/Intercom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Voicemail Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
System Info . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
G.729 Codec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
Backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
Update . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .134
Chapter 4
Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
Appendix A
Digium, Inc.
Page 10
Table Of Contents
Pin Assignments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Appendix B Specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
Appendix C Asterisk Appliance 50 (AA50) License Agreement . . . . . . . . . . . . 147
Appendix D Glossary and Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
Digium, Inc.
Page 11
List of Figures
Figure 1:
Figure 2:
Figure 3:
Figure 4:
Figure 5:
Figure 6:
Figure 7:
Figure 8:
Figure 9:
Figure 10:
Figure 11:
Figure 12:
Figure 13:
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Figure 19:
Figure 20:
Figure 21:
Figure 22:
Figure 23:
Figure 24:
Figure 25:
Figure 26:
Figure 27:
Figure 28:
Digium, Inc.
The Asterisk Appliance 50 (AA50) . . . . . . . . . . . . . . 17
Example Asterisk Appliance 50 Port Identification . . 22
Asterisk Appliance 50 Back View . . . . . . . . . . . . . . . 24
Mounting Instructions . . . . . . . . . . . . . . . . . . . . . . . . 27
GUI Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
System Status Page . . . . . . . . . . . . . . . . . . . . . . . . . 32
Configure Hardware . . . . . . . . . . . . . . . . . . . . . . . . . 35
Trunk Configuration Page . . . . . . . . . . . . . . . . . . . . . 40
New Analog Trunk Definition . . . . . . . . . . . . . . . . . . 41
Edit Analog Trunk Definition . . . . . . . . . . . . . . . . . . . 45
Add New Service Provider . . . . . . . . . . . . . . . . . . . . 47
Edit VoIP Service Provider . . . . . . . . . . . . . . . . . . . .48
Create New SIP/IAX Trunk Definition . . . . . . . . . . . . 50
Edit VoIP Trunk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
Outbound Calling Rules . . . . . . . . . . . . . . . . . . . . . . 55
New Calling Rule . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
Create New Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . 59
User Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Create New User . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
New Ring Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
New Call Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
Default Voice Menu . . . . . . . . . . . . . . . . . . . . . . . . . 78
Custom Voice Menu Prompts Page . . . . . . . . . . . . . 85
Record Menu Prompts . . . . . . . . . . . . . . . . . . . . . . .85
Upload Menu Prompts . . . . . . . . . . . . . . . . . . . . . . .86
New Time Interval . . . . . . . . . . . . . . . . . . . . . . . . . . .87
Incoming Calling Rules . . . . . . . . . . . . . . . . . . . . . . . 89
Page 12
List of Figures
Figure 29:
Figure 30:
Figure 31:
Figure 32:
Figure 33:
Figure 34:
Figure 35:
Figure 36:
Figure 37:
Figure 38:
Figure 39:
Figure 40:
Figure 41:
Figure 42:
Figure 43:
Figure 44:
Figure 45:
Figure 46:
Figure 47:
Figure 48:
Figure 49:
Figure 50:
Figure 51:
Figure 52:
Figure A-1:
Digium, Inc.
Incoming Calling Rules . . . . . . . . . . . . . . . . . . . . . . . 90
Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Paging/Intercom . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
New Page/Intercom Group . . . . . . . . . . . . . . . . . . . .98
Settings for Paging Individual Extensions . . . . . . . . . 99
Paging & Intercom Settings . . . . . . . . . . . . . . . . . . 100
New Conference Bridge . . . . . . . . . . . . . . . . . . . . . 101
Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .104
New Follow Me Definition . . . . . . . . . . . . . . . . . . . . 106
Follow Me Options . . . . . . . . . . . . . . . . . . . . . . . . . 108
Directory Settings . . . . . . . . . . . . . . . . . . . . . . . . . .110
Feature Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . .112
Call Parking Preferences . . . . . . . . . . . . . . . . . . . . 116
Application Map . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
Dial Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
New Voicemail Group . . . . . . . . . . . . . . . . . . . . . . . 122
System Information . . . . . . . . . . . . . . . . . . . . . . . . .123
Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
G.729 Codec Registration . . . . . . . . . . . . . . . . . . . 127
G.729 Codec License Information . . . . . . . . . . . . . 128
Backup Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
Asterisk Appliance 50 Update . . . . . . . . . . . . . . . . . 131
Polycom Update . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Asterisk Appliance 50 Options . . . . . . . . . . . . . . . .134
Back Panel Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Page 13
List of Tables
Table A-1:
Table A-2:
Table A-3:
Table B-4:
Digium, Inc.
CRAFT Port Pinout . . . . . . . . . . . . . . . . . . . .
RJ11 Analog Port Connector . . . . . . . . . . . . .
LAN & WAN Ethernet Port Pinouts . . . . . . . .
Maximum 12V Power Consumption . . . . . . .
. . . . . 142
. . . . . 143
. . . . . 144
. . . . . 146
Page 14
Chapter 1
Overview
®
The Digium Asterisk Appliance 50 (AA50) is a stand alone PBX which
runs Embedded Asterisk Business Edition™. It is suitable for the desktop,
or mounting in a typical network closet or restricted access location. The
Asterisk Appliance 50 is ideal for small office environments or as an
extension to a central Asterisk PBX.
The Asterisk Appliance 50 can function not only as a PBX, but also as a
voice mail server, IVR server, conferencing server, VoIP ATA, or VoIP
gateway. It has up to eight analog ports which are configured as FXO or
FXS ports depending upon the product model. Additionally, the built in
four port switch and WAN port allow it to also serve as a basic router.
The AsteriskGUI™ is the interface for the Asterisk Appliance 50. It gives
you the ability to configure the basic hardware and dial plan elements you
need when initially setting up your system, as well as every element
needed to customize your setup. You must create trunks, system users,
conferencing, voice mail, etc. The AsteriskGUI supports the following
browsers:

Firefox 1.5 through 3.0

IE 7

Safari 3.x

Opera 9.x
Digium, Inc.
Page 15
Chapter 1: Overview
Features:
 Embedded Asterisk Business Edition™
 AsteriskGUI™
 Four port 10/100BaseT Ethernet switch with Auto-MDI/MDI-X capa­
bility for the four 10/100BaseT LAN ports and one
10/100baseT WAN port (both 802.3/802.3u)
 Up to eight analog ports supporting either FXS or FXO lines depend­
ing on product version (available product versions: S800i with VoIP
only, S808i with Eight FXO, and S844i with Four FXS and Four FXO)
 SIP and IAX2 VoIP protocols
 CompactFlash interface (Type 1) suitable for standard CompactFlash
cards
 Configuration reset switch
 High performance Analog Devices Incorporated (ADI) BlackFin
BF537 processor
 uClinux Operating System
 Transcoding provided on the Blackfin processor
 32ms (Hardware Revision B) or 128ms (Hardware Revision C) of ana­
log port echo cancellation
 8MB on board serial Flash memory
 64MB 16 bit parallel SDRAM
 Front panel LEDs
Digium, Inc.
Page 16
Chapter 2
Unit Installation
This chapter provides the following information:
 Unpacking the Unit on page 18
 Inspecting Your Shipment on page 18
 Identifying Communication Ports on page 19
 Understanding the LEDs on page 19
 Using the Configuration Reset Switch on page 23
 Installing the Asterisk Appliance 50 on page 24
 Mounting the Asterisk Appliance 50 on page 27
Figure 1: The Asterisk Appliance 50 (AA50)
Digium, Inc.
Page 17
Chapter 2: Unit Installation
Unpacking the Unit
When you unpack your unit, carefully inspect it for any damage that may
have occurred during shipment. If damage is suspected, file a claim with
the carrier and contact your reseller from which the unit was purchased or
Digium Technical Support (+1.256.428.6161). Keep the original shipping
container to use for future shipment or proof of damage during shipment.
Note: Only qualified service personnel should install the unit. Users
should not attempt to perform this function themselves.
Inspecting Your Shipment
The following items are included in shipment of the Asterisk Appliance
50:
 Asterisk Appliance 50 (AA50)
 Compact Flash Card
 Power Supply
 Power Cable
 Analog Cables (optional depending on model)
 CD-ROM containing manual and installation files
 Product Registration Card
 Support and Warranty Information
Digium, Inc.
Page 18
Chapter 2: Unit Installation
Identifying Communication Ports
The Asterisk Appliance 50 unit consists of up to eight RJ11 analog ports
which are configured as FXO or FXS ports depending on the Asterisk
Appliance 50 model. These ports provide 32ms (Hardware Revisions B)
or 128ms (Hardware Revisions C) of analog port echo cancellation. The
unit is rated for a total of 8 REN across all FXS ports. Each individual
port is rated for up to 3 REN @ 1500ft (450m).
Four 10/100BaseT LAN ports and one 10/100BaseT WAN port provide
the functionality to connect to the local network as well as allowing the
Asterisk Appliance 50 to act as a router. All the Ethernet ports support
auto-MDI-X.
See Figure 2 on page 22 to locate the ports and their corresponding LEDs.
Understanding the LEDs
There are 15 LEDs on the front panel of the Asterisk Appliance 50. The
eight LEDs corresponding to the analog ports on the rear panel indicate
the type of interface installed. The definition of each LED and its color
representation is explained in Table 1.
Digium, Inc.
Page 19
Chapter 2: Unit Installation
Table 1: LED Definitions
LED
Color
Description
Power
Blue
(pulsing)
On when the unit boots up, after the
bootload process has completed. The LED
pulses at a rate which is proportional to the
processor load.
Compact
Flash
Blue
(flashing)
Flashes each time there is read or write
activity to or from the CompactFlash card.
WAN
Off
No line is connected or the interface is
inactive.
Green
(flashing)
Link is up at 100Mbps. LED flashes at 1/
10 second intervals as traffic is detected.
Orange
(flashing)
Link is up at 10Mbps. LED flashes at 1/10
second intervals as traffic is detected.
Off
No line is connected or the interface is
inactive.
Green
(flashing)
Link is up at 100Mbps. LED flashes at 1/
10 second intervals as traffic is detected.
Orange
(flashing)
Link is up at 10Mbps. LED flashes at 1/10
second intervals as traffic is detected.
LAN
(4 ports)
Digium, Inc.
Page 20
Chapter 2: Unit Installation
Table 1: LED Definitions
LED
Analog
(8 ports)
Digium, Inc.
Color
Description
Off
No analog port is installed in the
corresponding port.
Green
(solid)
Port is configured for FXS operation and
is enabled. An analog telephone may be
connected to this port.
Green
(flashing)
Telephone is ringing.
Green (slow
blinking)
Telephone is in use.
Red (solid)
Port is configured for FXO operation and
is enabled. A telephone line may be
connected to this port.
Red
(flashing)
Telephone line is ringing.
Red (slow
blinking)
Telephone line is in use.
Page 21
Chapter 2: Unit Installation
Figure 2: Example Asterisk Appliance 50 Port Identification
Digium, Inc.
Page 22
Chapter 2: Unit Installation
Using the Configuration Reset Switch
The Configuration Reset (CFG RST) switch (rear panel) will reset the
current Asterisk Appliance 50 configuration to the factory defaults when
pressed. The switch must be pressed and held during the boot process.
This will force the unit to delete all configuration data. The administrator
password will also be reset. See Figure 3 on page 24 to locate the CFG
RST switch.
Caution.
Pressing the CFG RST switch will cause loss of all
configuration settings and reset administration passwords.
Digium, Inc.
Page 23
Chapter 2: Unit Installation
Power
Supply
LAN
Craft
Ports
Port
Configuration
Analog
Reset Switch
WAN
Ports
Port
Figure 3: Asterisk Appliance 50 Back View
Installing the Asterisk Appliance 50
1. Remove the Compact Flash cover plate and insert the Compact Flash
card before connecting the power supply.
Caution.
The Compact Flash is not hot swappable. The Compact Flash
card should be inserted before powering on the unit. Likewise,
before removing the Compact Flash card it should be
unmounted (using the unmount command) and the Asterisk
Appliance 50 should be powered off.
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2. Connect one end of an Ethernet cable to an Asterisk Appliance 50
3.
4.
5.
6.
7.
8.
LAN port, and one end to an Ethernet connection on a computer
configured to obtain an IP address automatically (DHCP). This step
will connect your Asterisk Appliance 50 to your computer so that you
may access the Asterisk Appliance 50 GUI from your computer.
Connect the provided power cable to the power supply. You can then
connect the power supply to the Asterisk Appliance 50’s DC power
connector. The Asterisk Appliance 50 will immediately power on once
connected to a power source.
Using an Asterisk Appliance 50 supported web browser, open a
browser window and enter the IP address for the Asterisk Appliance
50. The default LAN IP address is 192.168.69.1. The default username
is admin, and the default password is password.
Note: The first time you log on you will be prompted to change
your password from the default. You will then be prompted to log
on with the new password. Once the log on process is complete the
AsteriskGUI home page will be displayed.
You may find it preferable to enable the Asterisk Appliance 50 GUI on
the WAN interface for ease of use. Once you have logged on to the
Asterisk Appliance 50, click on the Networking menu, and then the
WAN tab.
Select the Enable GUI on WAN interface checkbox.
Click Save, and then click Apply Changes. Your changes will be
applied and Asterisk will reload.
Attach the ethernet cable connected to the Asterisk Appliance 50’s
LAN port to the WAN port. Connect the other end of the cable to the
appropriate internet connection (will vary depending on your setup).
This will connect the Asterisk Appliance 50 to the internet.
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9. Connect telephones to the analog ports that are configured as FXS
ports and connect phone lines to the analog ports that are configured as
FXO ports.
10. Using an Asterisk Appliance 50 supported web browser, open a
browser window and enter the IP address for the Asterisk Appliance
50. The default username is admin, and the password is will be the
password you chose after first logging into the Asterisk Appliance 50.
11. You are now ready to configure your Asterisk Appliance 50 via the
GUI.
Caution.
This unit must be connected to the Telecommunications
Network in your country using an approved line cord, e.g.: for
Australia use only line cords complying with ACA Technical
Standard TS008.
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Mounting the Asterisk Appliance 50
Figure 4 below illustrates the proper mounting installation options:
Figure 4: Mounting Instructions
Warning.
Do not place anything (including paper) on top of the Asterisk
Appliance 50. To allow proper cooling, these units must not be
stacked.
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Table 2: Wall Mounting
Step
Instructions for Wall Mounting
1
Select the area to mount the Asterisk Appliance 50 unit
(refer to Figure 4 on page 27). The unit should be
mounted at or below eye level to properly view the
LEDs.
2
Install two #8 PAN headscrews (1 1/2-inch or longer)
into the desired location on the wall. They should be
placed approximately 7 1/2-inches, or 19cm, apart
horizontally or vertically, which is the distance between
the two keyed insets on the back of the Asterisk
Appliance 50. Make sure that the two screws are in
alignment and level.
3
Leave approximately 1/4-inch of the screw protruding
from the wall to allow the head of the screws to slide into
the keyed insets, mounting the unit to the wall.
Warning
The Asterisk Appliance 50 should not be mounted with the
LEDs pointing downward. Mounting the Asterisk Appliance 50
with the LEDs pointing downward may cause a disruption in
air circulation, which could cause the Asterisk Appliance 50 to
overheat. Mounting the Asterisk Appliance 50 this way can also
expose the LAN, WAN, and analog ports to potential damage.
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Chapter 3
Telephone System Configuration
This chapter provides information on how to initially set up your
telephone system via the AsteriskGUI™. The following topics are
covered:
 Log On to the Asterisk Appliance 50 on page 31
 The Asterisk Appliance 50 Interface on page 32
 Analog Hardware Configuration on page 35
 Trunk Configuration on page 40
 Outgoing Calling Rules on page 55
 Dial Plans on page 59
 User Extensions on page 61
 Ring Groups on page 68
 Music on Hold on page 70
 Call Queues on page 72
 Agent Login Settings on page 77
 Voice Menus on page 78
 Record a Voice Menu on page 85
 Time Intervals on page 87
 Incoming Calling Rules on page 89
 Voicemail on page 93
 Paging/Intercom on page 97
 Conferencing on page 101
 Follow Me on page 104
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








Directory on page 110
Call Features on page 112
Voicemail Groups on page 122
System Info on page 123
Networking on page 124
G.729 Codec on page 127
Backup on page 130
Update on page 131
Options on page 134
The Asterisk Appliance 50 comes with Embedded Asterisk Business
Edition™. The software includes the AsteriskGUI, a web based
configuration interface. The AsteriskGUI gives you the ability to set up
your telephone system without the need to use command line
configuration. After connecting to the Asterisk Appliance 50, the primary
menu is displayed, giving you the ability to configure your system, as
well as add features to your call system as your needs change.
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Log On to the Asterisk Appliance 50
Your Asterisk Appliance 50 should already be connected to an internet or
network connection, as described in Installing the Asterisk Appliance
50 on page 24. In the address field of an Asterisk Appliance 50 supported
web browser, enter the IP address assigned to your Asterisk Appliance 50.
The default LAN IP address is 192.168.69.1.
Figure 5: GUI Login
To log on to the system enter the following credentials:
 Username: admin
 Password: <password>
The first time you log on you will be prompted to change your password
from the default. You should have already chosen a new password during
the installation process. Once the log on process is complete the
AsteriskGUI home page will be displayed.
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The Asterisk Appliance 50 Interface
The AsteriskGUI gives you the ability to configure the basic hardware
and dial plan elements you need when initially setting up your system.
You must create trunks, system users, conferencing, voice mail, etc. After
logging into the AsteriskGUI, you’re presented with a variety of options
on the left side of the page.
Figure 6: System Status Page
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The AsteriskGUI supports the following browsers:

Firefox 1.5 through 3.0

IE 7

Safari 3.x

Opera 9.x
Every page of the GUI has two columns. The left column identifies all the
elements for which you can program the Asterisk Appliance 50. The
elements listed begin with System Status, which is the first page you see
upon logon, and proceed down to Options. Clicking any of the tabs on the
left of the page opens the corresponding page in the right column. Many
pages have additional information. Click on the information symbol, a
blue “i” enclosed in a circle, to get more information about a field or page.
The System Status page is the default page. This page shows you the
current version of firmware you are using, the status of any trunk lines
you have configured, the realtime status and additional details of all user
extensions, including the new and old voicemail message count for each
user extension (e.g. Messages: new/old), and the realtime status of all
agents, conference rooms, and parked calls. You can click on most
extension definitions to get more information. In addition, the System
Status page gives you the ability to log in, log out, pause, and unpause an
agent that is associated with one or more call queues.
Note: A user extension will have the status of “Unavailable” when the
VoIP account associated with it is not registered to the Asterisk
Appliance 50. The status will not change to “Unavailable” when a user
extension has both an analog port and a VoIP account associated with
it.
In the upper right corner of each page you will see the Apply Changes
and Logout buttons. Click Apply Changes to save and activate any
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changes you have made on a page so that you can utilize the changes.
Click Logout on any page to exit the Asterisk Appliance 50 GUI.
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Analog Hardware Configuration
You must configure your analog hardware according to the needs of your
system as part of your initial Asterisk Appliance 50 configuration. The
Configure Hardware page gives you the ability to configure both your
FXS and FXO ports, as well as your Tone Region, operation mode,
message waiting indicator mode (MWI), etc. The number of FXS and
FXO ports available for configuration will depend on the Asterisk
Appliance 50 model you purchased. Click the Configure Hardware tab
to configure your analog hardware.
Note: The Configure Hardware tab will not be available if you
ordered a VoIP only model.
Figure 7: Configure Hardware
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FXS and FXO ports provide the ability to receive and send calls through
the traditional telephone network, or POTS (Plain Old Telephone
System). FXS modules provide both dial tone and ringing voltage to an
analog phone. FXO modules accept dial tone and provide an interface to
the traditional phone lines. You plug a telephone line into an FXO port,
and an analog telphone into an FXS port.
On this page you can specify the signalling type for your FXS and FXO
ports. You have two choices; either Kewl Start or Loop Start. The Loop
Start method uses a short to request a dial tone. All North American home
phone lines use loop start signalling. Kewl Start is the same as Loop Start,
but is better able to detect disconnects. Select either Kewl Start or Loop
Start for each FXS and FXO module. Kewl Start is the default and is
preferred for analog circuits in Asterisk.
Note: Ground Start signalling is not supported.
You also need to select a tone region, which defines the set of tones (dial
tones, ringing tone, busy tone, etc) used in your region. Select your
country, or the nearest neighboring country, from the Tone Region dropdown list. The default setting is North America (United States/Canada).
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Advanced Analog Options
There are also some advanced settings which are applied to your analog
hardware. Specify them as needed, or accept the default values.
 Opermode - Setting operation mode, or Opermode, sets the On Hook
Speed, Ringer Impedance, Ringer Threshold, Current limiting, Tip/
Ring voltage adjustment, Minimum Operational Loop current, and and
AC Impedance selection as predefined for each countries analog line
characteristics. Select the country in which your Asterisk Appliance
50 is operating.
 A-law Override - Set the audio compression scheme. The setting you
choose is dependent on the country of operation. Ulaw is used in the
United States and Canada. A-law is used in most other countries. If
possible confirm the scheme which will be best for operation of your
Asterisk Appliance 50.
 FXS Honor Mode - This setting lets you choose whether you apply
the opermode setting to your FXO modules only, or to both FXS and
FXO modules.
 Boostringer - Set the voltage used for ringing an analog phone. Nor­
mal will set ring voltage to a normal level, or Peak will set the voltage
to 89v.
 Fast Ringer - The fast ringer tone can be set to normal, or to a 25hz
tone.
 Lowpower - The low power setting can be set to normal, or to a Fast
Ringer peak of 50v.
 Ring Detect - Users who are experiencing trouble detecting Caller ID
from Analog service providers or whose lines exhibit a polarity rever­
sal before Caller ID is transmitted from the provider should select Full
Wave. Otherwise, choose Standard.
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

MWI Mode - This option allows the user to specify the type of Mes­
sage Waiting Indicator detection to be done on trunk (FXO) interfaces.
The options are none, which performs no detection, FSK which per­
forms Frequency Shift Key detection, or NEON which perform Neon
MWI dection. The default value is none.
Echo Cancellation NLP Type - This option allows you to specify the
type of Non Linear Processor you want applied to the post echo-can­
celled audio reflections received from analog connections. There are
several options:
– None - This setting disables NLP processing and is not a recom­
mended setting. Under most circumstances, choosing None will
cause some residual echo.
– Mute - This setting causes the NLP to mute inbound audio streams
while a user connected to the appliance is speaking. For users in
quiet environments, Mute may be acceptable.
– Random Noise - This setting causes the NLP to inject random noise
to mask the echo reflection. For users in normal environments, Ran­
dom Noise may be acceptable.
– Hoth Noise - This setting causes the NLP to inject a low-end Gauss­
ian noise with a frequency spectrum similar to voice. For users in
normal environments, Hoth Noise may be acceptable.
– Suppression NLP - This setting causes the NLP to suppress echo
reflections by reducing the amplitude of their volume. Suppression
may be used in combination with the Echo cancellation NLP Max
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

Suppression option. For users in loud environments, Suppression
NLP may be the best option. This is the default setting for the Echo
Cancellation NLP Type option.
Echo Cancellation NLP Threshold - This option allows you to spec­
ify the threshold, in dB difference between the received audio (post
echo cancellation) and the transmitted audio, for when the NLP will
engage. The default setting is 24 dB.
Echo Cancellation NLP Max Suppression - This option, only func­
tional when the Echo Cancellation NLP Type option is set to Sup­
pression NLP, specifies the maximum amount of dB that the NLP
should attenuate the residual echo. Lower numbers mean that the NLP
will provide less suppression (the residual echo will sound louder).
Higher numbers, especially those approaching or equaling the Echo
Cancellation NLP Threshold option, will nearly mute the residual
echo. The default setting is 24 dB.
Note: The VPM Settings section will not be visible on older hardware
revisions of the Asterisk Appliance 50.
Once you have made the configuration changes to your hardware which
you require, click Save Changes. A message will display letting you
know that in order for these changes to be completed, you must reboot
your Asterisk Appliance 50. Click Options on the left menu, select the
Reboot tab, and then click Reboot Now to reboot your appliance.
Rebooting your Asterisk Appliance 50 will terminate any active calls.
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Trunk Configuration
Now that you have configured your analog hardware (assuming your unit
had any) you are ready to set up your trunk lines. Trunks are outbound
lines used to make calls. Trunks can be either analog or VoIP. Click
Trunks from the main menu to access the trunk configuration page.
Figure 8: Trunk Configuration Page
Trunk definitions are used in calling rules, dial plans, and call routing, etc.
You can use a mixture of both analog and VoIP trunks.
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Analog Trunks
Select the Analog Trunks tab to access the Manage Analog Trunks
page. Here you can create an analog trunk definition for each analog port
on your Asterisk Appliance 50. Click New Analog Trunk to open the
New Analog Trunk definition page.
Figure 9: New Analog Trunk Definition
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Use the following field definitions as a guide in creating your new analog
trunk definition.
 Channels - Select one or more analog channel (port) to be associated
with this trunk.
 Trunk Name - Specify a unique name to help you identify this trunk
when it is referred to in other areas such as calling rules.
 Busy Detection - This setting is used to detect far end hangup or for
detecting busy signal. Select Yes to enable this feature.
 Busy Count - If Busy Detection is enabled it is also possible to spec­
ify how many busy tones to wait for before hanging up. The default is
4, but better results may be achieved by setting to 6 or 8. The higher
the number, the longer it will take to hangup a channel. A higher num­
ber also lowers the possibility of false detections.
 Busy Pattern - If Busy Detection is enabled, it is also possible to
specify the cadence of your busy signal. In many countries it is 500
milliseconds on, 500 milliseconds off. Without Busy Pattern specified,
the Asterisk Appliance 50 will accept any regular sound-silence pat­
tern that repeats multiple times as a busy signal. If you specify Busy
Pattern, then the Asterisk Appliance 50 will check the length of the
sound (tone) and silence, which will further reduce the chance of a
false positive.
 Ring Timeout - Trunk (FXO) devices must have a timeout to deter­
mine if there was a hangup before the line was answered. This value
can be configured to shorten how long it takes before the Asterisk
Appliance 50 considers a non-ringing line to have hung up.
 Answer on Polarity Switch - If this option is enabled the reception of
a polarity reversal will mark when an outgoing call is answered by the
remote party.
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







Hangup on Polarity Switch - In some countries, a polarity reversal is
used to signal the disconnect (or hang up) on a phone line. If the
Hangup on Polarity Switch option is enabled, the call will be consid­
ered “hung up” on a polarity reversal.
Call Progress - On trunk interfaces it can be useful to follow the prog­
ress of a call through Ringing, Busy, and Answering. If turned on, Call
Progress attempts to determine answer, busy, and ringing on phone
lines. This feature is highly experimental and can easily detect false
answers and hang-ups. This may cause a hang up during the middle of
a call. Few zones are supported, but can be selected with the Progress
Zone option.
Progress Zone - This option defines the call progress zone for the
trunk interfaces.
Use CallerID - If this option is enabled Caller ID detection is also
enabled.
Caller ID Start - This option allows one to define the start of a caller
ID signal. Select Ring from the drop-down list to start caller ID when
a ring is received, or Polarity, to start caller ID when a polarity rever­
sal is detected.
Caller ID - This option allows the lines to report the caller ID string as
received from the telco, or as a fixed value by using the advanced
option.
Pulse Dial - If this option is enabled, pulse dialing, instead of DTMF,
will be used.
CID Signalling - This option defines the type of caller ID signalling to
use.
– bell - Bell202 as used in the United States
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– v23 - Used in the UK
– v23_jp - Used in Japan
– dtmf - Used in Denmark, Sweden, and Holland



Mailbox - This setting allows any message waiting indicator received
across the associated trunk to be forwarded to a local User, such as a
SIP phone.
Flash Timing - Flash Timing defines the duration, in milliseconds,
that Asterisk will use if it is sending a flash signal to another system.
Receive Flash Timing - Receive Flash Timing defines the duration, in
milliseconds, that Asterisk requires in order to consider a flash opera­
tion it receives to be valid.
Once you have completed the Analog Trunk definition, click Add. A
message will display letting you know that in order for these changes to
be completed, you must reboot your Asterisk Appliance 50. Before doing
so, you may wish to click the Edit button associated with an analog trunk
to configure additional options for tuning the audio.
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Figure 10: Edit Analog Trunk Definition
The Audio Tuning section will allow you to calibrate your analog ports
for optimum performance. Please ensure that your analog lines are
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plugged in before clicking the Easy Calibrate button. Your Asterisk
Appliance 50 must not have any active calls in order for the calibration
process to complete successfully on all analog ports. If you wish to reset
the calibration, click the Reset Calibration button.
Note: The Easy Calibration feature can take approximately 90 seconds
per port to complete.
In addition, an option to configure the gain level for each port will be
listed. This option can be used to raise or lower the audio level on your
ports. Normally, you should not have to adjust your analog ports beyond
the initial calibration. Should you still need to fine tune your audio
settings, please select one of the following:
 Low
 Soft
 Normal
 Loud
 Louder
Once you have completed the Analog Trunk definition, click Update. In
order for these changes to be completed, you must reboot your Asterisk
Appliance 50. Click Options on the left menu, select the Reboot tab, and
then click Reboot Now to reboot your appliance. Rebooting your
Asterisk Appliance 50 will terminate any active calls.
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Adding Service Providers
You must configure a VoIP service provider in order to connect to the
Public Switched Telephone Network (PSTN) via a VoIP connection.
Access to the PSTN gives you the ability to place calls to telephone
numbers no matter how they connect to the PSTN (VoIP or standard
analog system). Click the Service Providers tab to add a VoIP (SIP or
IAX) service provider.
Figure 11: Add New Service Provider
The list of VoIP service providers and corresponding configuration
information is pulled dynamically from a secure Digium webservice. If
you are already a VoIP provider customer, select the provider from the
list, click Add, and input your user name and password. Once you have
added a service provider it will appear in the Service Providers list. There
are Edit and Delete buttons associated with each Service Provider listing.
Click Edit to further refine your service provider definition. A detailed
definition will be displayed.
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Figure 12: Edit VoIP Service Provider
The Edit Service Provider page gives you the ability to change your
caller ID, as well as select a range of codecs.
 Username/Password - You will need to provide your log on creden­
tials in order to update your service provider information.
 Caller ID - The caller ID sent to the PSTN will be set to the value
specified in this field.
 Codecs - Codecs provide the ability for your voice to be converted to a
digital signal and transmitted across the Internet. The quality of your
call can be affected by the choice you make. The codecs available to
you will depend on what is supported by the service provider you
choose. You can select the order in which the codecs are used. The
codecs commonly available are u-law, a-law, GSM, G.726, G.722, and
G.729A. A registered G.729A license is required in order to use the
G.729A codec.
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Click Update when you have completed your changes, or Cancel to
discard your changes.
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Adding VoIP Trunks
If you do not have a subscription with one of the VoIP providers listed
above, or you have a special VoIP setup, you can add a custom VoIP
trunk. Click the VoIP Trunks tab to add a VoIP (SIP or IAX) service
provider. The Create New SIP/IAX Trunk page will be displayed.
Figure 13: Create New SIP/IAX Trunk Definition
Fill in the initial SIP/IAX trunk definition with the following information:
 Type - Select either the SIP or IAX protocol.
– SIP - Identifies that the trunk sends and receives calls using the
VoIP protocol SIP.
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– IAX - Identifies that the turnk sends and receives calls using the



VoIP protocol IAX.
Provider Name - Enter a unique name to help you identify this trunk
for use in calling rules, etc.
Hostname - The hostname or IP address assigned to the VoIP provider
or server.
Username/Password - You will need to provide your log on creden­
tials to the VoIP trunk server.
Note: If your VoIP trunk does not require a username, you may leave
the username field blank.
Click Add once you have completed your definition, or Cancel to discard
your changes.
Once you have added a VoIP trunk it will appear in the SIP/IAX trunks
list. There are Edit and Delete buttons associated with each VoIP trunk
listing. Click Edit to further refine your trunk definition.
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Figure 14: Edit VoIP Trunk
The following options will be available:
 Provider Name - Enter a unique name to help you identify this trunk
for use in calling rules, etc.
 Hostname - The hostname or IP address assigned to the VoIP provider
or server.
 Username/Password - You will need to provide your log on creden­
tials in order to update your service provider information.
 Codecs - Codecs provide the ability for your voice to be converted to a
digital signal and transmitted across the Internet. The quality of your
call can be affected by the choice you make. The codecs available to
you will depend on what is supported by the service provider you
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



choose. You can select the order in which the codecs are used. The
codecs commonly available are u-law, a-law, GSM, G.726, G.722, and
G.729A. A registered G.729A license is required in order to use the
G.729A codec.
Caller ID - This is the number the trunk will try to use when making
outbound calls. For some providers it is not possible to set the Call­
erID with this option. Thus this option may be ignored. When making
outbound calls the following rules are used to determine which Caller
ID is used, if they exist:
– The first Caller ID used is the Global CID defined in the Options
tab.
– The Caller ID set in the VoIP Trunks configuration, if defined,
takes precedence over the Global CID.
– The Caller ID set for the user making the call as defined in the
Users page will take precedence over the Global CID and the CID
set in VoIP trunks.
From Domain - If required by your provider, specify your primary
domain identity to show in the domain field of the From header for
outgoing SIP invites. Otherwise, only your IP address will be sent in
the From header.
From User - If required by your provider, specify the user to show in
the user field of the From header for outgoing SIP invites. Otherwise,
only your IP address will be sent in the From header.
Insecure - This is a SIP parameter used to determine peer matching.
The setting determines whether or not an insecure connection will be
allowed, or if authentication is required. The valid options are:
– port - Enter this value to match against only an IP address. This set­
ting is useful if you have multiple endpoints behind a NAT device.
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– very - Specify this value if you do not want to require authentication

upon an initial invite.
– no - Specify this value if you do not want to allow an insecure con­
nection.
Enable Remote MWI - When you select this option, you enable
voicemail from your remote provider. Typically a user’s voicemail is
stored locally on the Asterisk Appliance 50. The notification of new
voice mail is provided by the same local Asterisk Appliance 50. If you
would like to receive voicemail notifications from a remote provider,
this option is available. To enable this option, click the check box, and
in the Remote Mail Box field, specify the remote mail box number or
identity to which you wish to subscribe, e.g. 6001. Select the local user
who should receive this MWI notification. Please note: enabling this
option for a local user will disable the local user’s Asterisk Appliance
50 voice mail. It is not possible to provide local voice mail and remote
MWI simultaneously.
Click Add when you have completed your changes, or Cancel to discard
your changes.
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Outgoing Calling Rules
An outgoing calling rule pairs an extension pattern with a trunk used to
dial the pattern. This allows different patterns to be dialed through
different trunks (e.g. "local" 7-digit dials through an analog line but "long
distance" 10-digit dials through a low-cost SIP trunk). You can optionally
set a failover trunk to use when the primary trunk fails. The Outgoing
Calling Rules give you the ability to use basic pattern matching to
differentiate outbound calls and route them accordingly. The tab displays
each outgoing calling rule established and the service providers assigned.
Figure 15: Outbound Calling Rules
Note: Outbound Calling Rules manages only individual outgoing
call rules. See the Dial Plans section to associate multiple outgoing
calling rules to be used for User outbound dialing.
The Calling Rules menu shows every rule name established, the pattern
the rule will match against, the trunk used to complete the call, and the
failover trunk to be used. of call. Several default calling rules will be
available when you initially set up your Asterisk Appliance 50. Click on
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Add a Calling Rule to define a new calling rule. The following dialog
will be displayed.
Figure 16: New Calling Rule
A calling rule is comprised of the following items:
 Calling Rule Name - Choose a name that describes the type of rule
you are creating, e.g. “Local” or “Long Distance”.
 Pattern - The Pattern field gives you the ability to use basic pattern
matching to differentiate calls and route them accordingly. For
instance, if a number begins with _9256, and is followed by 7 or more
digits, that would define a call within the state of Alabama. If a call
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bly doesn’t require a long distance charge. Instead of adding a rule for
every extension or phone number you call, specify the pattern in this
rule similar to the example. All patterns begin with the underscore “_”
character. There are special characters which can be used in patterns:
– X - Any digit from 0-9
– Z - Any digit from 1-9
– N - Any digit from 2-9
– [1,2,3,6-9] - Any digit within the brackets, in this instance 1, 2, 3, 6,
7, 8, 9.
– . - The period is the wildcard which will match anything remaining.
For example, _9011. matches anything starting with 9011.
– ! - The exclamation point is a wildcard which causes the matching
process to complete as soon as it can determine that no other
matches are possible.
Send to Local Destination - Calls matching the pattern specified will
be routed to the destination specified in Destination if this checkbox
is selected.
Destination - Specify a destination, such as voicemail or main menu,
for calls to be routed to when the Send to Local Destination checkbox is selected.
Use Trunk - Specify the trunk through which calls, matching the spec­
ified pattern, will be placed.
Strip - This option gives you the ability to remove specified number of
digits from the front of the call string before the call is dialed and
placed through the trunk specified in Use Trunk.
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Prepend These Digits - This option gives you the opportunity to add
digits to the front of the call string before the call is dialed and placed
through the fail over trunk. For example, a 3 digit area code could be
prepended to a 7 digit string for calls to a service provider which
requires 10 digit dialing.
Note: You may also prepend the ‘w’ character for analog trunks to
provide a 500ms delay before dialing. This is useful if your
telecommunications provider does not immediately provide dial tone
after going off hook.
Use Failover Trunk - Failover trunks can be used to ensure that a call
goes through if the primary trunk is busy or down. If the Use Failover
Trunk checkbox is selected and Fail Over Trunk is specified, then
calls that can not be placed through the primary trunk will be placed
through this alternate route. If your primary trunk is a VoIP trunk, but
you want calls to be placed through the PSTN when the VoIP trunk
isn’t available, then this option will suit your needs.
Once you have completed the calling rule definition click Save to accept
the rule or Cancel to abandon your changes. Click Apply Changes in the
upper right corner of the page to make your changes immediately
available. Click Edit next to a rule on the calling rule list to edit a
previously defined rule, or Delete to delete the rule.
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Dial Plans
A Dial Plan is a collection of Outgoing Calling Rules. Dial Plans are
assigned to user extensions to specify the dialing permissions associated
with that extension. For example, you might have one Dial Plan for local
calling that only permits extensions associated with that Dial Plan to dial
local numbers, via the "local" outgoing calling rule. Another extension
may be permitted to dial long distance numbers, and so would have a Dial
Plan that includes both the "local" and "longdistance" outgoing calling
rules.
Click New at the top of the Calling Rules page and create a new dial plan
name. You can then add calling rules for that dial plan definition.
Figure 17: Create New Dial Plan
The default dial plan, the collection of your calling rules, is
Default_Dialplan. You can create more than one dial plan, especially if
you want to have different dial plans for different user extensions. Change
the DialPlanName, and then select the checkbox for each Outgoing
Calling Rule associated with this plan. You can also select which local
contexts, such as conferences, voicemenu, and queues should be part of
the dial plan.
Once you have completed the dial plan definition click Save to accept the
plan, or Cancel to abandon your changes. Click Apply Changes in the
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upper right corner of the page to make your changes immediately
available. Click Edit next to a dial plan on the list list to edit a previously
defined plan, or Delete to delete a dial plan.
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User Extensions
The User Extensions page is used to create individual user accounts on
the system. Each user definition includes an extension, name, password,
etc. User extension definitions are the basic components of your phone
system. They are needed for voicemail, conferencing, call queues, dial
plans, etc. Click the Users tab to view the main User Extensions page.
Figure 18: User Extensions
The main page lists all previously created user extensions. You can edit
individual users as well as change attributes of several users at the same
time. Your first step when setting up a new system will be to create one or
more users. Click Create New User to create a new user extension.
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Figure 19: Create New User
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Extension - The numbered extension, e.g. 6000, assigned to the
defined user. The extension must be a number within the range speci­
fied in Extension Preferences on the Options page.
Name - The first and last name of the individual assigned to this
extension. The name can also be that of a department, such as Sales or
Support, for example. This is important because the Dial By Name
Directory function of Asterisk uses this information to route calls.
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Dial Plan - This option references the Dial Plans option on the left
tool bar. Based on the calling rules you’ve created, you can restrict the
outbound dialing of this extension to local calls, emergency calls, and
standard long-distance calls for North America. This option also possi­
bly allows blocking or allowing international (011 prefix dialed) calls.
Caller ID - Identifies the Caller ID presented when the listed exten­
sion dials an internal extension.
Outbound Caller ID - Identifies the Caller ID presented when the
listed extension dials an extenal number. Your ability to manipulate
your outbound CID may be limited by your VoIP provider. Manipula­
tion of CID across analog trunks is not possible.
Enable Voicemail - Builds a voice mail box for the extension that can
be reached by dialing the Check Voicemail extension. The Voicemail
extension can be configured. The current default is 6050.
Voice Mail Access Pin Code - The password used to access voicemail
for the specified extension.
E-mail Address - Voice mails received by this extension can be sent
as audio file attachments e-mailed to a specific address.
SIP - Identifies whether the extension sends and receives calls using
the VoIP protocol SIP.
IAX - Identifies whether the extension sends and receives calls using
the VoIP protocol IAX.
Analog Station - A drop-down menu is available to identify the ana­
log phone port which this extension will access. If more than one
phone is connected to your Asterisk Appliance 50 you may need to
confirm the port number listed on the back of the Asterisk Appliance
50.
Flash - Flash Timing defines the duration, in milliseconds, that Aster­
isk will use if it is sending a flash signal to another system.
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RXFlash - Receive Flash Time defines the duration, in milliseconds,
that Asterisk requires in order to consider a flash operation that it
receives to be valid.
Codec Preference - Codecs provide the ability for your voice to be
converted to a digital signal and transmitted across the Internet. The
quality of your call can be affected by the choice you make. The
codecs available to you will depend on what is supported by the ser­
vice provider you choose. You can select the order in which the codecs
are used. The codecs commonly available are u-law, a-law, GSM,
G.726, G.722, and G.729A. A registered G.729A license is required in
order to use the G.729A codec.
MAC Address - The MAC Address field is used to specify the MAC
address of a PolyCom® phone connected to the Asterisk Appliance 50.
The MAC address associates the phone with this extension and
enables the auto-synchronization of provisioning information.
Line Number - Polycom brand VoIP phones are capable of servicing
1 to 6 separate VoIP phone lines, depending on the model of the
phone. If you are using the Polycom Auto-provisioning feature of the
Asterisk Appliance 50, this option can be used to define which line of
your phone will be used by the user. No more than one user can be
assigned to a line on a phone.
Note: Each phone must be configured with a user that has Line
Number set to 1. Additionally, assigned line numbers must be in a
contiguous range.
Line Keys - Polycom brand VoIP phones include multiple line keys.
The number of line keys available will depend on the model of the
phone. If you are using the Polycom Auto-provisioning feature of the
Asterisk Appliance 50, this option can be used to define how many
line keys on the phone should be associated with this user (e.g. Let’s
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says you configure a single Polycom phone with two users. User 6000
with Line Number set to 1 and Line Key set to 2 will display user 6000
on line keys 1 and 2 on the phone. User 6001 with the same MAC,
Line Number set to 2, and Line Key set to 4 will display user 6001 on
line keys 3, 4, 5 and 6 on the phone.). Be sure not to select more line
keys than your phone supports.
SIP/IAX Password - The password used if the user has a SIP/IAX
account.
NAT - Try this setting when your Asterisk Appliance 50 is on a public
IP, communicating with devices behind a NAT device (broadband
router). If you have one-way audio problems, you usually have prob­
lems with your NAT configuration or your firewall's configuration of
SIP and RTP ports.
Can Reinvite - By default, the Asterisk Appliance 50 will route the
media streams from SIP endpoints through itself. Enabling this option
causes the Asterisk Appliance 50 to attempt to negotiate the endpoints
to route the media stream directly. It is not always possible for the
Asterisk Appliance 50 to negotiate endpoint-to-endpoint media rout­
ing. This option can be used to tell the Asterisk Appliance 50 whether
or not to issue a reinvite to the client.
DTMF Mode - Set the default DTMF mode for sending DTMF (touch
tone). The default setting is rfc2833. Other options include:
– info - Used to display SIP Info messages
– inband - Inband audio (requires 64 kbit codec - alaw, ulaw)
– auto - Use rfc2833 if offered, inband otherwise.
Insecure - Insecure is a SIP parameter used to determine peer match­
ing. The setting determines whether or not an insecure connection will
be allowed, or if authentication is required. The valid options are:
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– port - Enter this value to match against only an IP address. This set­
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ting is useful if you have multiple endpoints behind a NAT device.
– invite - Enter this value to match against both the IP address and
port number provided in the Contact field of the SIP header. A call
will be allowed without authentication if a match is found.
– very - Specify this value if you do not want to require authentication
upon an initial invite.
– no - Specify this value if you do not want to allow an insecure con­
nection.
3-Way Calling - Allows the extension to receive a call and then dial
out to another phone number to conference with the inbound call and
the recipient of the outbound call.
In Directory - Check this option if you want a user to be searchable
using the system telephone directory.
Call Waiting - If call waiting is not enabled, the extension accepts
only one call before it is identified as busy.
CTI - Selecting this option (Computer Telephony Integration) allows
the user to connect applications to the Asterisk Management Interface.
Is Agent - Call queuing is made up of a bank of agents who receive
calls. An extension listed as Is Agent can be added to queues from the
Call Queues option.
Pickup Group - A Pickup Group is a group of user extensions. Each
member of a pickup group can answer another member’s phone by
dialing *8. Select the pickup group to associate with the user exten­
sion.
Once you have completed the user extension definition click Save to
accept the definition, or Cancel to abandon your changes. Click Apply
Changes in the upper right corner of the page to make your changes
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immediately available. Click Edit next to a user extension on the list to
edit a previously defined extension, or Delete to delete the user definition.
Editing Multiple User Definitions
You can edit multiple user definitions by selecting one or more user’s
checkboxes and then click Modify Selected Users. You will be able to
edit the definition attributes common to all users such as Dial Plan,
voicemail PIN, or Pickup Group setting. Click Update to update the
selected users, or Cancel to abandon your changes. You can also delete
multiple users by selecting one or more users from the displayed list and
clicking Delete Selected Users. Click OK to complete the deletion, or
Cancel.
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Ring Groups
Ring groups allow a group of phones, or devices, to ring simultaneously
or in sequence (ring order). This provides the opportunity for multiple
people to answer a call (ring all) or one person can answer a call from any
phone. The Asterisk Appliance 50 does not come with a default ring
group. To create a new ring group click New Ring Group at the top of the
Ring Groups page.
Figure 20: New Ring Group
Note: You need at least one member to be able to define a ring group.
You will not be able to define a ring group without any user
extensions.
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To create a ring group, use the following procedure.
1. Define the Name of the group. The name can be any mnemonic such
as Sales or Technical Support.
2. Specify an extension to associate with the ring group. This is the
extension that can be dialed to ring all members of the group
simultaneously or in order of listing.
3.
4.
5.
6.
Note: Go to Options, General Preferences to see which range of
numbers have been specified for ring groups.
Choose the members of the ring group from the Available Users list.
Click on a user extension or trunk, and then click the arrow pointed at
the Ring Group Members list to transfer. Select a user extension or
trunk in the Ring Group Members list and then click the arrow
pointing toward Available Users to transfer the selected item back to
the list. Click the double arrow symbol to transfer all group members
back to the Available Users list.
Choose a ring group strategy from the Strategy drop-down list. You
can choose either Ring All which will ring all phones in the defined
group simultaneously, or Ring Order which will ring phones in
sequence determined by the order of the users or trunks in the group.
Specify the number of seconds that each phone (or all phones) should
ring before ringing the next phone in order.
Lastly, determine which action you want the system to take if no one
answers the call. You can either direct the call to the voicemail of a
user, go to an IVR menu, or end the call.
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Music on Hold
Music on hold is the music played to individuals on hold or during
conference calls while conference members are waiting for the call to
begin. The Asterisk Appliance 50 comes with a default group, or class, of
sound files which can be used for music on hold. Click Music on Hold
and then select the default class to see the list of default sound files.
Figure 21: Music on Hold
If you think the default music is acceptable, but you’d like to give your
system a more customized feel, you can also upload your own music or
sound files. Each file uploaded must be less than 10 megabytes, in 8KHz
mono, and in ulaw, alaw, g722, or gsm format. Not sure how to convert
your audio to an acceptable format? Linux users should try the Sox utility,
and Windows users should look into Audacity. Any conversion program
is acceptable as long as the file meets the upload criteria.
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Click New MOH Class to create a new label for a new group of music on
hold files. Select the newly created class from the Music on Hold list, and
then use the upload form to upload new music on hold files to the list.
Once you have uploaded your files, click Apply Changes to make the
files available. You can now use them for call queues, parked calls,
conferences, etc.
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Call Queues
A call queue lines up callers and allows them to wait to speak to any
group of employees taking a high volume of calls. The feature allows you
to speak to more people rather than send callers back to voice mail to
leave a message and receive a call back when time permits.
Asterisk identifies which extensions under the Users tab are capable of
belonging to a call queue by whether the Is Agent option is selected. The
Is Agent option indicates that the user is available to answer customer
calls. If a check mark does not appear next to Is Agent, that extension
won’t appear in the list of agents in the configuration for this option.
Figure 22: New Call Queue
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The Call Queues page, with the Queues tab selected, lists the existing
queues. None will be listed if you have not yet created a queue. To create
a new queue, click Create New Queue. Use the following steps to create
a queue. Keep in mind the purpose of the queue and how it should
operate.
Creating a Queue
1. The extension for the queue will automatically populate in the Queue
field with the next available extension. If you want the number to be
something other than the automatically chosen one, enter it in the
Queue field.
Note: Go to Options, General Preferences to see which range of
numbers have been specified for ring groups.
2. Next, give the queue a name that will be meaningful. The queue will
be referenced by this name, so be sure to make it sufficiently
descriptive as well. For example, “Technical Support” for the technical
support queue, “Sales”, and so on.
3. You now should choose the strategy used in your queue call logic.
Using the Strategy drop-down list, choose one of the following
options for routing calls:
– Ring All - Rings every agent who isn’t on an active call when a new
call arrives. The first agent to answer the call receives it.
– Round Robin - Every available agent receives a call in turn, akin to
how cards are dealt in a poker game.
– Least Recent - The agent who has been without a call the longest
receives the next call.
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– Fewest Calls - The agent who has handled the fewest calls receives
the next incoming call.
– Random - Goes by the luck of the draw; any agent can receive the
next incoming call.
– RrMemory - This option is Round Robin with Memory. It’s similar
to Round Robin, but smarter — it remembers over the course of
days, weeks, or years which agent received the last call so that it can
commence with the next agent in sequence when calls begin again.
4. The Agents box lists all Users that are designated as an agent that can
receive calls as part of a call queue. All users listed have the Is Agent
checkbox selected on their user profile. Many Users may be listed as
potential agents, but some may be assigned to a sales queue and some
for a service queue. This box lists all agents and enables you to choose
which users you assign to the queue.
You have now filled in the basic information necessary to create a call
queue. There are additional queue options available to control the timing
and managing of the calls, as well as the agents. You may not want to
work with these finer points of call queuing until after your call queue has
been working for a while, and you have an idea of call volume and the
turnover of calls by each agent.
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Music on Hold - Select the music on hold class to associate with this
call queue. Music on hold can be managed on the Music on Hold
page.
Join Empty - This option allows callers to enter a queue even if no
agents are logged into it. There are three options available:
– Yes - Callers can join a queue with no agents or only unavailable
agents.
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– No - Callers can not join a queue with no agents. This is the default
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option.
– Strict - Callers can not join a queue with no agents or if all agents
are unavailable.
Leave When Empty - This option mirrors the Join Empty, but it rep­
resents a queue in which agents had been logged in but are now gone.
At 5:00 pm, when your employees go home, you can program the
queue to shut down when the agents log out. The existing callers in
queue are forced to exit, and no new callers are granted access to the
queue. There are three options available:
– Yes - Callers are forced out of a queue when no agents are logged in.
– No - Callers will remain in a queue with no agents.
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– Strict - Callers are forced out of a queue with no agents logged in,
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or if all agents logged in are unavailable. This is the default option.
Timeout - The default for this option is 15, representing 15 seconds
that an agent’s phone will ring before the call is forwarded on to
another agent.
Wrapup Time - This is a buffer of time allowing your agents to finish
work on one call and remain unavailable in the queue. The default on
this option is 0 seconds, providing no buffer time for your agents and
allowing the next call to ring through immediately after a call is com­
plete.
Max Len - This option sets the maximum number of callers allowed
in the queue before they are sent to voice mail or receive a busy signal.
The default is “0,” which allows for an unlimited number of calls in
queue before they are sent elsewhere.
Auto Fill - This option allows multiple calls that arrive at the same
time to be immediately forwarded on to agents.
Auto Pause - If an agent fails to answer a call, this option temporarily
postpones sending calls to that agent.
Report Hold Time - The Report Hold Time tells the agent how long
the call was holding in queue before it was sent to the agent. If the hold
time was short, the agent will probably be happy to accept the call. If
the hold time was 10, 15, or 20 minutes, the agent might want to brace
for a frustrated customer, but at least the agent isn’t overwhelmed.
Click Update to add the new queue, or Cancel to abandon your changes.
Once saved the new queue will be displayed on the Manage Queues
page. You can edit or delete any previously created queue from the
Manage Queues page.
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Agent Login Settings
The Agent Login Settings tab, accessible from the Manage Queues
page, lets you specify the extensions for agents to log into their queue, as
well as callback login.
 Agent Login Extension - Use this field to specify the extension which
all agents can dial to log into the queue(s) associated with their exten­
sion.
 Agent Callback Login Extension - Use this field to specify the exten­
sion which all agents can dial to log into the queue(s) associated with
their extension. This is the same as Agent Login, but the agent does
not have to remain on the line.
 Agent Logout - To logout of Agent Login just hang up your phone.
To logout of Agent Callback Login, dial the same extension used to
login, specify your extension and password when prompted, and press
# when asked for your callback extension. This will successfully log
you out of all queues.
Click Save to retain the agent login settings.
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Voice Menus
A valuable feature of Asterisk is the ability to create Interactive Voice
Response (IVR) or voice menus. Voice menus are designed to allow for
more efficient call routing. The menus provide a caller with specific
instructions, receive responses from the caller, and process those
responses into an action.
Each Asterisk Appliance 50 ships with a default voice menu already
created. To better understand the creation and operation of these menus,
we will examine the default one.
Figure 23: Default Voice Menu
Voice menus are constructed depending on your needs. Just like your
business you need to create the solution best suited to your customers.
The best way to understand how a voice menu is constructed is to
examine the default “Welcome” menu provided with your Asterisk
Appliance 50. Click Voice Menus - Welcome in the Voice Menus list.
The options for the welcome menu are displayed similar to the example
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shown in the above illustration. The Welcome menu consists of the
following steps:
 Answer the Call
 Wait ‘1’ Sec
 Play ‘thank-you-for-calling & Listen for KeyPress
 Play ‘if-u-know-ext-dial’ & Listen for KeyPress
 Play ‘otherwise’ & Listen for KeyPress
 Play ‘to-reach-operator’ & Listen for KeyPress
 Play ‘pls-hold-while-try’ & Listen for KeyPress
 WaitExten ‘6’ Sec
In the example, when a caller dials your company number ending in 7000,
the call is answered, and after a pause of one second the caller is greeted
in the following manner: “Thank you for calling. If you know your party’s
extension, please dial it now. Otherwise to reach an operator please dial
0.” If the caller tries an extension, the menu will respond with “Please
wait while I try that extension.” If no action is taken by the caller, the
menu will repeat after 6 seconds.
This is an example of a basic voice menu. In the example, each action is a
step chosen from the list of available menu options. The available menu
options are as follows:
 Answer - This step is automatically added when creating a new menu.
This step answers the incoming call.
 Authenticate - The Authenticate step is used to restrict access to one
or more areas of your system. This is useful when one wants users to
have to enter a PIN code in order to proceed to a particular part of the
current menu, to a different menu, or to ring an extension.
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Background - This step is used to play an audio file in the background
while waiting for the caller to enter an extension or number. Playback
is terminated once the user begins to enter an extension. To select a file
to play, click and hold in the field next to the Background choice to
scroll through a list of pre-recorded sound files. In the example above,
“Play ‘otherwise’ & Listen for KeyPress” is an example of using the
Background option.
Busytone - The Busytone option should be selected if there is a step in
the process in which you want to play a busy signal to the caller. You
would play the busyone to the caller, for instance, if the call is over.
Congestion - The Congestion option is similar to the Busytone
option. The Congestion option should be selected if there is a step in
the process in which you want to play a congestion signal to the caller.
You would play the congestion signal to the caller, for instance, if you
want to indicate the line is not available.
Digit Timeout - The Digit Timeout option is used to set the maximum
amount of time allowed between key presses. If a full extension is not
entered in the specified time, the entry will be considered invalid. A
field for entering the number of seconds before timeout appears next
to the option.
DISA - DISA (Direct Inward System Access) is an application which
allows callers from outside the system to get access to an internal dial
tone and place calls from within your internal system. A passcode is
required. If the passcode entered is correct, the user is given a system
dial tone on which a call may be placed.
Note: Use caution when choosing this option. This option can pose a
security risk.
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Response Timeout - If a caller does not enter a response with the time
specified in this field, the call will terminate. This step could be put at
the end of a series of menu choices.
Playback - The Playback option is similar to the Background option
because it will play a sound file you select. However, this option does
not allow interruption from a KeyPress event, and will move on to the
next step in your list.
Set Music on Hold Class - Set the group of music on hold files to be
associated with this voice menu.
Wait - The Wait option pauses the execution of steps in the voice
menu list for the number of seconds you specify.
WaitExten - The WaitExten option is specified to give a caller a
specified amount of time to enter an extension.
Goto Destination - The Goto Destination option lets a caller choose
to go to either a voice menu, a specific extension, voicemail box, or a
ring group from a list of destinations.
Set Language - This option gives you the ability to set the language
for voice prompts in your voice menu. This option is especially useful
if you want to begin with the default language, and then give the
option of setting a different language for the rest of the menu. For
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example, voice prompts will begin in English, but if a user is given a
choice, and presses 2 for Spanish, all further voice menu prompts will
be in Spanish (provided that language module is loaded).
Goto Directory - The Goto Directory option sends a caller to the sys­
tem phone directory. This gives the user the chance to select a user
name from the directory if the extension is unknown.
Dial a Number via Trunk - This option allows you to specify an
external number to dial, including the trunk that should be used for the
call.
User Event - This option gives you the ability to send an arbitrary
event to the manager interface, with an optional body representing
additional arguments. Specify the eventname in the User Event field.
If necessary, specify additional arguments in the Body field.
Hangup - The Hangup option terminates the call.
Custom App - This option allows you to specify an Asterisk applica­
tion, along with the application’s corresponding parameters, which is
not already listed in the Add new Step drop-down menu (e.g.
‘NoOp(hello world)’ to echo “hello world” on the Asterisk CLI).
Note: The Custom App option is only visible when Advanced
Options are enabled under the Options menu item. This option
should only be configured by experienced Asterisk administrators.
Refer to section titled Advanced Options on page 138 for further
details.
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Creating a Voice Menu
Use the following procedure as a guide to creating a voice menu.
1. On the Voice Menu page, click New to create a new voice menu.
2. Specify a Name and an Extension. The extension will be the direct
dial to the voice menu.
3. Specify the Steps of your voice menu using the welcome menu
example and step descriptions as guides.
4. Select the Dial Other Extensions checkbox if you want to give a user
the ability to break out of the menu selections and dial an extension
within your system.
Warning: The Dial Other Extensions option is important. This option
allows an inbound caller to break out of the listed Keypress Events and
dial another extension. A malicious person may be able to hack
through your Asterisk implementation to find an outside dial tone and
use it for fraud. Any extensions that are known to the public should be
completely handled by the Keypress Events; callers should not be
allowed to dial other extensions. Sticking to this policy protects your
Asterisk system from being compromised.
5. Specify the Keypress Event actions for digits 0-9 as well as *, #, t,
and i. The options available for a Keypress Event are:
– None - The associated key is not enabled.
– Goto Menu - Pressing a key with this option will send the caller to a
specified menu.
– Goto Extension - Pressing a key with this option will send the
caller to a specified extension.
– Goto Queue - Pressing a key with this option will send the caller to
the specified queue.
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– Operator - This option will send the caller to the designated opera­
tor.
– Hangup - Pressing a key with this option will terminate the call.
– Congestion - Pressing a key with this option will play a busy signal.
– Both the t key and i key should be used for specific actions. The
action associated with the t key should be the desired action if a user
response has timed-out. The action associated with the i key should
be the desired action if a user makes an invalid entry.
6. Once you have constructed your voice menu, click Save. You can then
click Apply Changes to add the voice menu to your current
configuration.
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Record a Voice Menu
In the event that one wants to record custom menu prompts for Asterisk
which can be used in a voice menu, the Voice Menu Prompts tab may be
used.
Figure 24: Custom Voice Menu Prompts Page
A list of previously recorded menus is displayed on the Custom Voice
Menu Prompts page. Here, the user may modify several options:
 Record Again - Clicking this button allows the user to make another
attempt at recording and replacing an existing custom sound file.
 Play - Clicking this button brings up a dialog entry box to allow the
input of an extension that Asterisk will dial and play the prompt.
 Delete - Clicking this button will delete the selected prompt.
Figure 25: Record Menu Prompts
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Click Record a new Voice Menu Prompt to record a custom voice menu
prompt. The following options will be available:
 File Name - This text entry box specifies the saved name of the file
that is to be recorded.
 Format - Select whether the recording will be in GSM or WAV for­
mat.
 Extension Used for Recording - This drop-down select box allows
the user to choose which extension Asterisk will dial to wait for the
user to speak the prompt.
 Record - Clicking this button causes Asterisk to launch the call that
will record a file.
Figure 26: Upload Menu Prompts
Click Upload a Voice Menu prompt to upload a custom voice menu
prompt. You will be prompted to specify the path to the audio file that you
wish to upload. Each file uploaded must be less than 10 megabytes, in
8KHz mono, and in GSM or WAV format.
Once your recording or upload of a custom voice menu prompt is
finished, it will be listed on the Custom Voice Menu Prompts page. You
will be able to play back the prompt, re-record the prompt, or delete the
prompt. The prompts can now be included when creating voice menus.
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Time Intervals
Time intervals are definitions of a period of time during a day, week,
month, etc. which are used to route calls. Time interval definitions are
utilized in the Incoming Calling Rules section. To define a time interval,
select Time Intervals from the left menu, and then New Time Interval
from the Time Intervals page.
Figure 27: New Time Interval
Creating a Time Interval definition is fairly simple. You just need to
define a range of time in which you expect to receive calls. The following
fields are used to create the definition:
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



Time Interval Name - Specify a unique name to help you identify this
time interval when it is referred to in the creation of calling rules. A
name can be anything such as BusinessHours, OffHours, or Holiday.
By Day of Week - Select this radial button if you wish to specify one
or more days of any week. Select the range of days using the dropdown lists. For example, if you were creating the time interval “Busi­
ness Hours” you would specify Monday in the first drop-down list and
Friday in the second drop-down list. For time intervals that occur on a
single day, select that day in both drop-down lists.
By Days of a Month - Select this radial button if you wish to specify a
day of a specific month instead of a day of a week. Enter the day of the
month, and then select the month from the drop-down list. For exam­
ple, if you were creating a time interval named Christmas, you would
enter “25” and then select “December” from the drop-down list.
Time - You need to specify a time during which this interval should be
applied. Select either the Entire Day checkbox, or a Start Time and
End Time. In the Business Hours example, which is from Monday to
Friday, you would specify a start time of 8:00 AM and an end time of
5:00 PM. In the “Christmas” example you would select the Entire
Day checkbox.
Click Update to save your time interval definition, or Cancel to discard
your changes. Click Apply Changes to make the new time interval
active.
Once a time interval definition is created, you can either Edit or Delete
the definition from the Time Interval page.
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Incoming Calling Rules
Incoming Calling Rules give you the ability to use basic pattern matching
to route incoming calls based on time intervals for each analog or VoIP
trunk with which you receive inbound calls. Click Incoming Calling
Rules to access the Incoming Calling Rules page.
Figure 28: Incoming Calling Rules
The main page displays the incoming calling rules created for each trunk.
No rules are displayed if you have just setup your Asterisk Appliance 50.
Click New Incoming Rule to create a new incoming calling rule. The
new incoming rule form will be displayed.
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Figure 29: Incoming Calling Rules
There are only a few options you will need to define to create a new rule.



Trunk - Select the trunk which the incoming rule should apply to from
the drop-down list. The trunk can be either an analog or VoIP trunk.
Time Interval - Select the time interval from the list available in the
drop-down list. You may have created time intervals for business
hours, weekend hours, holiday time, etc. You can also select None if
you want to bypass any time intervals or patterns.
Pattern - The Pattern field gives you the ability to use basic pattern
matching to differentiate calls and route them accordingly. For
instance, if a number begins with _9256, and is followed by 7 or more
digits, that would define a call within the state of Alabama. If a call
began with _9 followed by 7 digits, it would be a local call that proba­
bly doesn’t require a long distance charge. Instead of adding a rule for
every extension or phone number you call, specify the pattern in this
rule similar to the example. All patterns begin with the underscore “_”
character. There are special characters which can be used in patterns:
– X - Any digit from 0-9
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– Z - Any digit from 1-9
– N - Any digit from 2-9
– [1,2,3,6-9] - Any digit within the brackets, in this instance 1, 2, 3, 6,
7, 8, 9.
– . - The period is the wildcard which will match anything remaining.
For example, _9011. matches anything starting with 9011.
– ! - The exclamation point is a wildcard which causes the matching
process to complete as soon as it can determine that no other
matches are possible.

Note: If you have selected an analog trunk, this field will be grayed
and populate with an s. This is not a pattern, but an indication that the
analog phone should proceed to the destination.
Destination - Select the Destination for the incoming call. You can
choose to send the call to to either a voice menu, a specific extension,
voicemail box, ring group, the operator, or even hang up the call.
– The Local Extension by DID destination setting allows you to
route the incoming call to a local extension based on the DID
(Direct Inward Dialing) number that is sent to you by your telecom­
munications provider. Upon selecting Local Extension by DID,
you will notice the Local Extension by DID Pattern option appear.
This option gives you the ability to remove a specified number of
digits from the front of the DID number string before routing the
call to a local extension.
Note: The Local Extension by DID destination setting is not applicable for analog trunks.
The rules you need to create are dependent on your needs. If you are
configuring your system for a business, for example, you’ll probably want
to set up rules for business hours, off hours, weekend hours, etc. In any
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case, you should also create a calling rule which utilizes the time interval
and uses a catch all pattern to route any calls that don’t fit the other rules
you’ve created. This will insure that you don’t miss any calls.
Once you have completed the definition of each incoming calling rule,
click Update. Click Apply Changes in the upper right corner of the page
to make your changes immediately available. Each rule you create will be
listed on the Incoming Calling Rules page, organized by trunk. From the
main page you can either Edit or Delete the rule.
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Voicemail
Voicemail is an option available for every extension. The relationship
between the extension and voicemail is established in Users. In that
section you can specify whether voicemail is enabled for an extension, as
well as the PIN for retrieving voicemail. The Voicemail page lets you
specify voicemail parameters, as well as settings for sending voicemail
notices to e-mail.
Figure 30: Voicemail
There are three tabs on the Voicemail page used for configuration:
General Settings, Email Settings, and SMTP Settings.
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General Settings
The General Settings page is the primary page used to configure Asterisk
Appliance 50 voicemail. Standard configuration information is present,
allowing you to confirm the extension used to check messages, as well as
general parameters such as the following:
 Extension for Checking Messages - This option defines the exten­
sion which Users call in order to access their voicemail account.
 Direct Voicemail Dial - Select this checkbox to enable direct voicemail dialing. For example, someone would be able to dial *6001 to
directly dial the voicemail box and leave a message for the person at
extension 6001 if this checkbox is selected.
 Max Greeting (Seconds) - With this option, you specify the maxi­
mum amount of time available to record your voicemail greeting.
 Dial “0” for Operator - Callers who are sent to voicemail can press
“0” for the operator and be transferred either during the voicemail sal­
utation, or after recording the message. If this option is not enabled, a
caller’s pressing “0” will be ignored.
There are several options which can define the characteristics of the
voicemail messages in the system.
 Maximum Messages per Folder - This field sets the maximum num­
ber of messages that a user can have in any over their voice mail box
folders.
 Maximum Message Time - The maximum duration of a message left
by a caller. Time is specified in seconds.
 Minimum Message Time - The minimum duration of a message
specified in seconds. Any message left that’s under the listed duration
is discarded and isn’t processed or retrievable.
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There are also several message playback options which can be specified.
 Say Message Caller-ID - The Say Message Caller ID option reads the
caller ID before the voice mail message is played.
 Say Message Duration - If this option is enabled the duration of the
message, in minutes, will be played back before the voicemail mes­
sage is played.
 Play Envelope - Turn on/off playing introductions about each mes­
sage when accessing them from the voicemail application.
 Allow Users to Review - This option provides incoming callers the
option to review their message before it is saved and can be played
back by the owner of the voicemail extension. Standard options are
presented to the caller, allowing them to discard the message or re­
record it.
E-mail Settings
The E-mail Settings page is used to set e-mail options for voicemail, as
well as the format of the e-mails sent.



Note: SMTP settings must be specified in order to send e-mail.
Send Messages by E-mail Only - If this option is set, voicemail mes­
sages will only be accessible by e-mail.
Attach Recordings to E-Mail - This option is used to choose whether
voicemail is sent to a users e-mail address as an attachment. Click the
check box to enable this option. Messages will be sent in the .wav for­
mat.
Template for E-mails - The e-mail template gives you the ability to
specify the general content for each e-mail sent with a voicemail alert.
To load a sample template, click the Load Defaults button. Be sure to
change the From address to a valid e-mail address before saving.
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SMTP Settings
The SMTP Settings page is used to enable sending voicemail alerts
through e-mail.
 SMTP Sever - The IP address or a hostname of an SMTP server
which the Asterisk Appliance 50 can connect to, without authentica­
tion, to send voicemail notifications to an e-mail address.
 Port - The port number on which the SMTP server is running. The
default port is 25.
 Use SMTP Authentication - Click this checkbox if the SMTP server
requires a username and password for authentication.
 Auth User - The username used for authentication to the SMTP
server.
 Auth Password - The password used for authentication to the SMTP
server.
Once you have completed specifying the settings on a tab, click Save to
keep your settings, or Cancel to discard your settings. Click Apply
Changes in the upper right corner of the page to make your changes
immediately available.
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Paging/Intercom
The Paging/Intercom tab allows you to set up 1-way paging or 2-way
intercom for calling an individual or a group of extensions. This can be
used to make an announcement over the speakerphone of a group of
phones. Phones which are part of a page/intercom group will not ring, but
will immediately answer into speakerphone mode.
Note: This functionality is dependent on a compatible and correctly
configured handset. For a user to be able to dial a page/intercom
group, the ‘pagegroups’ local context must be included in the user’s
dialplan.
Figure 31: Paging/Intercom
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Click New Page/Intercom Group to define which available users will be
part of a page/intercom group.
Figure 32: New Page/Intercom Group
The following options are available when defining a new page/intercom
group:
 Extension for this Page/Intercom Group - Specify the extension
associated with this page/intercom group.
 Type - Specify the type of group for this extension.
– 2-Way Intercom - The person initiating the call and all members of
the intercom group will be able to speak to each other during the
call.
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– 1-Way Page - Only the person initiating the call will be able to



speak during the call. All members of the paging group will be
muted.
Play a beep - If this option is checked, a beep sound will be played
when the intercom call is connected to inform users that they can
begin talking.
Page/Intercom Group Members - This is the list of available users
which are part of this page/intercom group.
Available users - This is the list of users which are available to be
assigned to this page/intercom group.
The double left arrows will move all available users to this page/intercom
group. The double right arrows will remove all page/intercom group
members. The single left arrow will be move an individual available user
to the page/intercom group. The single right arrow will remove an
individual page/intercom group member.
Click Save to retain your page/intercom group, or Cancel to abandon
your changes. From the Paging & Intercom page, you can either Edit or
Delete a page/intercom group.
Click Page an Extension along the top to configure a key sequence
which initiates a page or intercom call to a specific extension.
Figure 33: Settings for Paging Individual Extensions
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The following settings are available:
 Prefix for Paging an Extension - Specify the key sequence used to
prefix a page call to a specific extension. For example, setting this
value to ** would allow you to initiate a page call to extension 6000
by dialing **6000.
 Prefix for Dialing an Extension as intercom - Specify the key
sequence used to prefix an intercom call to a specific extension, For
example, setting this value to *# would allow you to initiate an inter­
com call to extension 6000 by dialing *#6000.
Click Save to retain your changes, or Cancel to abandon them.
Then click Settings along the top to specify additional settings for paging
and intercom.
Figure 34: Paging & Intercom Settings
The following setting is available:
 Alert-Info Header - This is the value that is sent in the alert info
header to the phone for an intercom call. It is not recommended that
this value be changed from the default of Intercom.
Click Save to retain your changes, or Cancel to abandon them. Once you
have completed making changes to the Paging & Intercom sections,
click Apply Changes to make them immediately available.
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Conferencing
Every company reaches the point of needing more people on a phone call
than it can effectively include through three-way calling. Conference
bridges allow you to include more people as well as project a professional
image. The configuration of the conference bridge and standard features
is very straightforward. Click New Conference Bridge on the
Conferencing page to design a conference bridge.
Figure 35: New Conference Bridge
The GUI auto-populates the extension with the next available extension in
sequence, but can be changed to any extension number that is available.
After establishing the extension for the bridge, you need to specify the
password settings for the conference. Assign the PIN Code used by
participants to enter the conference as well as the Administrator PIN
Code used by the moderator of the conference to open the conference
bridge.
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Now that you have established the conference bridge extension and
password codes, you can set your conference room options.
 Marked/Admin User Extension - This option works in conjunction
with the Wait for Marked User feature. If the conference bridge is to
have marked users or admin users, those users should enter the confer­
ence from a separate extension. Admin users can lock and unlock the
conference, and can kick the most recent conference participant.
Marked users are special users whose entrance and exit, if the Wait
for Marked User or Close Conference When Last Marked User
Exits are selected, can either begin or end the conference. If the CEO
of the company, for example, doesn’t want anyone chatting in the con­
ference bridge until he or she arrives, these options are set to keep
everything quiet. The main conference extension of 6003 is configured
with Wait for Marked User selected. Everyone in the conference arriv­
ing from extension 6003 remains silent until the CEO arrives.
 Play hold music for first caller - Checking this option makes music
play for the first caller entering a conference until another caller joins.
Some people don’t like sitting in a quiet room — even a virtual room
— alone, and this feature prevents anyone from being in that position.
 Enable caller menu - This feature allows callers to access the Confer­
ence Bridge Menu by pressing the asterisk (*) key.
 Quiet Mode - Do not play enter/leave sounds when callers join or
leave the conference.
 Close Conference When Last Marked User Exits - When this
option is selected, the conference call will be closed when the last
marked user exits the call.
 Announce callers - All new callers to a conference are identified
when they arrive when this feature is selected.
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
Wait for Marked User - This is a feature that keeps all participants in
quiet mode until a special participant, using a unique extension,
arrives. Only after the marked user arrives is the audio activated so
that all of the participants can speak to each other.
Click Update to retain your conference bridge definition, or Cancel to
abandon your changes. From the Conferencing page you can either Edit
or Delete a bridge definition. Once you have saved a conference bridge
definition click Apply Changes to make the bridge immediately
available.
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Follow Me
Follow Me is a very useful feature which allows a caller to reach you
wherever you may be by forwarding your calls to a list of predefined
numbers until you are reached. If you cannot be reached, Follow Me will
transfer the caller to your voicemail box. The Follow Me feature may also
be referred to as Find Me.
Note: The Follow Me feature will only function for user extensions
which have voicemail enabled.
Figure 36: Follow Me
The following is an example scenario of using the Follow Me feature:
1. Derrick dials extension 6000 from his mobile phone to call Chuck.
2. Chuck’s office phone rings several times, but is not answered.
3. Derrick hears, “After the tone, say your name, and then press the
pound key.”, followed by a beep tone.
4. Derrick says his name, and then presses the pound key.
– If Derrick had not said his name and/or pressed the pound key, the
call would have continued on to the next step as normal.
5. Derrick hears, “Thank you. Please hold while I try to locate the person
you are calling.”
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6. Then Chuck’s mobile phone and home phone begin to ring
simultaneously.
7. Chuck’s android at home answers the phone and hears, "Incoming call
from”. Then it hears Derrick state his name. Then it hears, “Press 1 to
accept this call, or 2 to reject it.”
– This occurs while Chuck’s mobile phone continues to ring.
8. Chuck’s android quickly hangs up the phone instead of pressing 1 or 2.
– If Chuck’s android had pressed 1, it would have begun speaking
with Derrick, and Chuck’s mobile phone would have stopped ring­
ing.
– If Chuck’s android had pressed 2, the call would have been rejected,
Chuck’s mobile phone would have stopped ringing, and Derrick
would have been transferred to Chuck’s voicemail box.
– If Chuck’s android had not hung up the phone and not pressed any­
thing, the message would have looped itself until either the ring tim­
eout for the Follow Me number had been met or Chuck had
answered his mobile phone, whichever would have come first.
– If the ring timeout for the Follow Me number had been met while
Chuck’s android was listening to the accept/reject message, it would
have been disconnected from the call, Chuck’s mobile phone would
have stopped ringing, and Derrick would have been transferred to
Chuck’s voicemail box.
– If Chuck would have answered and accepted the call from his
mobile phone while his android was listening to the accept/reject
message, it would have been disconnected from the call, and Chuck
would have begun speaking with Derrick.
9. Chuck answers his mobile phone and hears, "Incoming call from”.
Then he hears Derrick state his name. Then he hears, “Press 1 to
accept this call, or 2 to reject it.”
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10. Chuck presses 1 to accept the call.
11. Lastly, Chuck begins speaking with Derrick.
Note: If no one had answered and accepted the call, Derrick would
have been transferred to Chuck’s voicemail box.
Figure 37: New Follow Me Definition
Use the following procedure as a guide to configure Follow Me for an
user extension.
1. Click Edit for the user extension which you wish to configure. The
edit box for the Follow Me definition will appear.
2. In order to enable the Follow Me feature, select Enable for the Status
option.
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3. Select the ‘Music On Hold’ Class which you would like for the caller
to hear while Follow Me attempts to reach you.
4. Select the DialPlan that should be used for dialing the Follow Me
numbers. The dial plan associated with the user extension will be
selected by default.
5. Click the Add Follow Me Number button in order to create a list of
Follow Me numbers which will be dialed to reach the user. Upon
doing so, additional options will appear near the bottom of the edit
box.
6. Select Dial Local Extension if you would like to specify a local
extension on the system to be dialed, or Dial Outside Number if you
would like to specify an outside number to be dialed.
Note: In order to properly match one of the patterns in your Outgoing
Calling Rules, be sure to prepend the necessary digits when specyfing
the outside number to be dialed.
7. Specify the number of seconds before the ring timeout occurs for the
new Follow Me number. The ring timeout for the new Follow Me
number is the total amount of time from when the Follow Me feature
is initiated to when the call is accepted.
Note: The ring timeout is not reset or cancelled when the prompt is
played to allow someone to accept or reject the call. If the ring timeout
is met while that prompt is being played, the call will be rejected and
sent to voicemail.
8. Select the Dial Order in which this Follow Me number should be
dialed to reach the user.
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– Selecting Ring after trying previous extension/number will cause
the defined Follow Me number to be called after the last entry listed
in the Destinations box.
Note: You must select Ring after trying previous extension/
number if no other Follow Me number exists in the Destinations
box. Otherwise, you will be unable to save the Follow Me
definition.
– Selecting Ring along with previous extension/number will cause
the defined Follow Me number to be called simultaneously along
with the last entry listed in the Destinations box.
9. Click Add to add this Follow Me number to the Destinations box, or
Cancel to discard it.
You may reorder the entries in the Destinations box by using the up and
down arrows located to the far right of each entry. If you wish to delete an
entry, simply click the X located next to the up arrow.
Click Save to retain your changes or Cancel to discard them.
Then click FollowMe Options along the top to configure additional
options for Follow Me.
Figure 38: Follow Me Options
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The following self-explanatory options can be enabled or disabled:
 Playback the incoming status message prior to starting the follow-me
step(s).
 Record the caller’s name so it can be announced to the callee on each
step.
 Playback the unreachable status message if we’ve run out of steps to
reach the callee, or if the callee has elected not to be reachable.
Click Save to retain your changes or Cancel to discard them. Then click
Apply Changes to make the changes available.
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Directory
The Directory settings page gives you the ability to set your preferences
for the Dial by Names Directory. Dialing the directory extension gives
callers the opportunity to search the telephone directory by first or last
name.
Figure 39: Directory Settings
On this page you specify the extension for dialing the system directory, as
well as announcement and search preferences.
 Directory Extension - The extension to dial to access the names
directory.
 Also read the extension number - Select this checkbox if you would
like the extension number as well as user name to be read before pre­
senting dialing options to the caller.
 Use first name instead of last name - Select this checkbox if you
want to give callers the ability to search on first name instead of last
name.
Click Save to retain your changes or Cancel to discard them. Click Apply
Changes to make the changes available. To add or remove a user from the
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system telephone directory, edit the In Directory field of user’s extension
accessible from the Users page.
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Call Features
The Call Features tab gives you the ability to configure feature codes,
call parking, application maps, and dial options. These are explained in
the following sections.
Feature Codes
The Feature Codes tab gives you the ability to define a keypress
sequence which will initiate a blind transfer, attended transfer, call park,
or call disconnect.
Note: Feature codes will only function when two channels are
answered and bridged together. They cannot be used while the remote
party is ringing or in progress.
Figure 40: Feature Codes
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The checkbox must be selected for any feature for which you wish to
define a custom key sequence. The feature code options are described
below.
Note: Take care when specifying the key sequence for each feature
code. The key sequence detection will stop as soon as it finds a
possible match (e.g. If you have the key sequence for Blind Transfer
set to ‘#’ and Attended Transfer set to ‘#2’, pressing ‘#2’ during a call
will initiate a blind transfer instead of an attended transfer because
DTMF detection will stop after pressing ‘#’. An example of properly
configuring two feature codes starting with ‘#’ would be to set the key
sequence for Blind Transfer to ‘#1’ and Attended Transfer to ‘#2’.)
Blind Transfer Feature Code
Specify the key sequence to initiate the Blind Transfer feature during an
active call. The default key sequence is ‘#’. Blind Transfer may also be
referred to as an unannounced, unsupervised, or cold transfer.
When initiated, this feature will prompt you to enter the destination
extension for the blind transfer. You must then enter the destination
extension within a few seconds, otherwise the blind transfer will be
cancelled. After entering the destination extension within the alloted time,
the calling party will be transferred to the destination extension without
prior notification and the initiator of the transfer will be disconnected.
The calling party’s Caller ID will be preserved when the call is transferred
to the destination extension.
Note: The T Option and/or t Option must be enabled under Dial
Options in order for this to function.
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Disconnect Feature Code
Specify the key sequence to initiate the Disconnect feature during an
active call. The default key sequence is ‘*’.
When initiated, this feature will disconnect the active call.
Note: The H Option and/or h Option must be enabled under the Dial
Options tab in order for this to function.
Attended Transfer Feature Code
Specify the key sequence to initiate the Attended Transfer feature during
an active call. A default key sequence is not defined for this feature.
Attended Transfer may also be referred to as an announced, supervised,
consult, full-consult, or warm transfer.
Initiating this feature will prompt you to enter the destination extension
for the attended transfer. You must enter the destination extension within
a few seconds, otherwise the attended transfer will be cancelled. After
entering the destination extension within the alloted time, you will hear
ringback if the destination extension is available. If the destination
extension answers, you will be given the opportunity to announce the call
transfer. Simply hang up the phone to complete the call transfer. If you
hang up before the destination extension answers, the calling party will be
transferred to the destination extension without prior notification (i.e.
similar to blind transfer, but without CallerID preservation). If the
destination extension does not answer and you do not hang up the phone,
the attended transfer will be cancelled after 15 seconds. The calling
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party’s Caller ID will not be preserved when the call is transferred to the
destination extension.
Note: The T Option and/or t Option must be enabled under the Dial
Options tab in order for this to function.
Call Parking Feature Code
Specify the key sequence to initiate the Call Parking feature during an
active call. A default key sequence is not defined for this feature.
Initiating this feature will prompt you with the first available parking
extension. This is the number that can be dialed to retrieve the call from
the parking lot. The caller will be immediately transferred to the specified
parking extension, and the initiator of the call park will be disconnected.
In order to retrieve the call, dial the parking extension that was specified
by the Asterisk Appliance 50 prompt.
The amount of time that the call remains parked is determined by the
number of seconds specified in the Number of seconds a call can be
parked for field on the Call Parking tab. If the call is not retrieved
within this time, the call will be redirected to the user that originally
parked the call.
Note: The K Option and/or k Option must be enabled under the Dial
Options tab in order for this to function. This method of Call Parking
may also be referred to as one step parking.
Click Save when you are done configuring this section. Then click Apply
Changes to make these changes immediately available for new calls.
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Call Parking
Call Parking is an Asterisk feature which allows a user to place a call on
hold so that it can be taken off hold from another extension. Click the Call
Parking tab from the Call Features page to configure this feature. The
Call Parking page gives you the ability to define the call parking options
which will enable use of this feature.
Figure 41: Call Parking Preferences
The following options must be configured to enable call parking.
Extension to Dial to Park a Call - Specify the extension to call when
transferring a call to hold or the “parking lot”.
 What Extensions to Park Calls On - The extensions specified here
will be the “parking lot” designations for the calls you place on hold.
The call on hold will be retrieved by dialing one of these extensions.
 Number of Seconds a Call Can Be Parked - The number of seconds
a call can be placed on hold. After the time has elapsed the call will
ring the originating extension.

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Parking a Call
You can park a call using either an analog or VoIP phone. To use an
analog phone, hit the flash button, or quickly press the hook switch, wait
for a dial tone, then dial the extension (700). With a VoIP phone, initiate
the transfer, dial the call parking extension (e.g. 700), then complete the
transfer (such as by pressing send). The method using a VoIP phone will
vary depending on the phone.
At this point, the Asterisk Appliance 50 will prompt you with a number.
The number it prompts you with is the number from the pool specified.
This is the number that can be entered to retrieve the call. To retrieve the
call, pickup a phone, and dial the parking number that was previously
specified by the Asterisk Appliance 50 prompt. The amount of time that
the call remains parked is determined by the number of seconds specified.
If the call is not retrieved in this time, the call will be redirected to the
user that originally parked the call.
Note: In order to properly park a call, you must use attended transfer
functions. Using a blind transfer function will not provide the parking
number to the person parking the call. This makes recovery of the call
impossible, except for the fall through timeout.
Application Map
The Application Map tab gives you the ability to define a keypress
sequence which will execute a specific application, along with the
application’s arguments. One example of using this feature would be to
allow the caller or callee to playback a specific sound file on demand
when pressing a predefined key sequence. Click New Application Map
to define a new application map.
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Figure 42: Application Map
The options associated with an application map are described below.
 Enabled - Select whether or not this application map is enabled.
 Feature Name - Specify a unique name to be associated with this
application map.
 Digits - Specify the key sequence used to activate this feature.
 Activate On / By - Select which channel of the call that the applica­
tion will be executed on, and which channel is allowed to activate this
feature. The available settings are describe below.
– self - Run the application on the same channel that activated this
feature. This feature will be accessible by both the caller and callee.
– peer - Run the application on the opposite channel from the one that
has activated this feature. This feature will be accessible by both the
caller and callee.
– self / caller - Run the application on the same channel that activated
this feature. This feature will be accessible by the caller only.
– peer / caller - Run the application on the opposite channel from the
one that has activated this feature. This feature will be accessible by
the caller only.
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– self / callee - Run the application on the same channel that activated


this feature. This feature will be accessible by the callee only.
– peer / callee - Run the application on the opposite channel from the
one that has activated this feature. This feature will be accessible by
the callee only.
– self / both - Refer to the description for self.
– peer / both - Refer to the description for peer.
App Name - Select the application to execute once the defined key
sequence is detected.
Note: The application map feature is not intended to be used for all
Asterisk applications. Applications which are statically defined in
the extensions.conf are executed by the PBX core. In contrast,
applications which are dynamically called from an application map
are executed outside of the PBX core. It is not appropriate to use
any application which has any concept of dialplan flow when using
an application map. Examples of this would be applications such as
Macro, Goto, Background, and WaitExten.
Arguments - Specify the arguments to be passed to the application
defined in App Name.
Note: Enabling the application map feature will cause the Asterisk
Appliance 50 to remain in the media stream during all calls. This will
occur regardless of whether two endpoints are configured to redirect
their media stream from the Asterisk Appliance 50 to each other after
the call setup has completed (e.g. two SIP phones with reinvite
enabled).
Click Save when you are done configuring this section. Then click Apply
Changes to make these changes immediately available for new calls.
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Dial Options
The Dial Options tab gives you the ability to configure feature code
permissions for the called party and the calling party. Either party can be
allowed or restricted access to the transfer, hang up, and call parking
feature codes.
Figure 43: Dial Options
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These permission options are explained below.
 t Option - Allows the called party to transfer the calling party by send­
ing the DTMF sequence defined on the Feature Codes page.
 T Option - Allows the calling party to transfer the called party by
sending the DTMF sequence defined on the Feature Codes page.
 h Option - Allows the called party to hang up by sending the DTMF
sequence defined on the Feature Codes page.
 H Option - Allows the calling party to hang up by sending the DTMF
sequence defined on the Feature Codes page.
 k Option - Allows the called party to enable parking of the call by
sending the DTMF sequence defined on the Feature Codes page.
 K Option - Allows the calling party to enable parking of the call by
sending the DTMF sequence defined on the Feature Codes page.
Click Save when you are done configuring this section. Then click Apply
Changes to make these changes immediately available for new calls.
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Voicemail Groups
A voicemail group gives you the ability to create a voicemail box that can
be shared by any of the users on an Asterisk Appliance 50 system. A
group message can thus be sent by dialing one extension and leaving a
message. Click Voicemail Group to access the Voicemail Group page.
Figure 44: New Voicemail Group
Click New Voicemail Group to create a voicemail group.
Voicemail Group Extension - Specify the group voicemail extension.
 Label - Specify a unique name for the voicemail group which can be
referred to in the configuration of your Asterisk Appliance 50.
 User Mailboxes - Click the checkbox of each user voicemail box
which should be part of the group voicemail box.

Click Save when you have completed the voicemail group definition, and
then Apply Changes to make the voicemail box immediately available.
You can either Edit or Delete the voicemail group from the main
Voicemail Group page.
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System Info
The general system information of the Asterisk Appliance 50 is displayed
from this tab, as well as tabs for your network interfaces, disk usage,
memory usage, and DHCP leases allocated on the LAN side by the
Asterisk Appliance 50’s DHCP server. If you do not have an NTP server
specified, you can set your default time zone from within the General tab.
The S800i Config tab describes the exact model information. This
information will be useful for technical support.
Figure 45: System Information
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Networking
The Networking page is used to configure your general network settings,
as well as your Wide Area Network (WAN) settings, Local Area Network
(LAN) settings, and Timezone settings.
Figure 46: Networking
The General tab, which is the default selection on the Networking page, is
used to specify the following settings:


Hostname - The hostname assigned to the Asterisk Appliance 50.
This name will be used to identify the Asterisk Appliance 50 on your
network.
NTP Server - This field gives you the ability to specify the URL or IP
address of an NTP server. This is useful if you wish to regularly syn­
chronize the Asterisk Appliance 50 time setting with that of an NTP
server.
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

SSH - Select the SSH checkbox to activate the SSH server on the
Asterisk Appliance 50. The default root password is digium. Enabling
this option will cause your unit to provide SSH access on both WAN
and LAN interfaces, which can pose a security risk.
Note: It is suggested that you change the default root password the
first time you SSH into the Asterisk Appliance 50. Use the “passwd”
utility from the shell to change the password. Changing the default
password will increase security.
URL for Auto-Provisioning - The URL specified in this field is used
to enable auto-provisioning for Polycom phones. The default for this
field is http://0.0.0.0/phoneprov. The 0.0.0.0 will resolve to the IP of
the Asterisk Appliance 50 with the LAN IP for requests from the LAN
ports, and the WAN IP for requests over the WAN port.
The WAN tab is used to specify the settings which will enable connection
to the Internet, or to an internal, private network.
 DHCP - The DHCP setting enables the automatic assignment of an IP
address to the Asterisk Appliance 50. This checkbox is selected by
default.
 Enable GUI on WAN Interface - Select this checkbox only if you are
certain you want to enable access to the Asterisk Appliance 50 GUI
via the WAN interface.
 Enable WAN Side Provisioning - Select this option to enable the pro­
visioning of Polycom phones connected through the WAN.
If you have difficulty obtaining an IP address dynamically, deselect the
DHCP checkbox and specify the IP address, Subnet, Gateway, and DNS
settings. This information should be available from your company
network administrator or Internet Service Provider (ISP).
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The LAN tab is used to specify the settings for your local network. A
local network is usually a smaller network which is part of a WAN. The
information specified here is used to access your Asterisk Appliance 50.
The default IP address specified, 192.168.69.1, is used to access the
AsteriskGUI. You can change this address to an IP address specified in
the IP start and end ranges. In most cases, the default should be used.
The Timezone tab is used to specify the default timezone for your
Asterisk Appliance 50. The time zone information is used to set the date
and time on the Asterisk Appliance 50. The time zone files are located on
the flash card which comes with your Asterisk Appliance 50. Select the
appropriate timezone from the list. Click Set as Default to set the
corresponding time zone as your default time zone. Clicking the Update
Timezones button will download and install the latest timezone files from
Digium’s website. You will need to restart your Asterisk Appliance 50 in
order to complete setting the time zone as your default. To reboot your
appliance, go to Options, Reboot, and click Reboot Now.
Note: The time zone files are named after cities that adhere to the time
zone you need.
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G.729 Codec
The G.729 Codec page allows you to register and manage your G.729
codec license keys. The G.729 Codec is an industry standard algorithm
that compresses and decompresses a digital audio stream. Applied to
Voice over IP (VoIP) calls, G.729 compresses the audio data to use
significantly less network bandwidth than a standard or uncompressed
VoIP call. This compression allows for more calls to be carried without
increasing network capacity and allows voice to travel on limitedbandwidth connections that would otherwise not support VoIP.
Note: The Asterisk Appliance 50 can support a maximum of 8
simultaneous calls being transcoded with G.729. Calls using G.729
which are not being transcoded (pass-thru) do not count towards that
total.
Figure 47: G.729 Codec Registration
Click Register a G729 License key on this Appliance to cause the
Asterisk Appliance 50 to download the End User License Agreement
(EULA) from a server at Digium. Read the EULA carefully. If you agree
to its terms, click I agree to the above License.
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Figure 48: G.729 Codec License Information
Complete the G.729 Codec License Information form in full. The G.729
License Key field should begin with “G729-”. Then click Register. Your
registration information will be securely sent to Digium’s registration
server.
The registration process may take a few minutes to complete. A message
box will appear to let you know once the firmware has finished updating.
You must reboot your Asterisk Appliance 50 in order for these changes to
take effect. Click Options on the left menu, select the Reboot tab, and
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then click Reboot Now to reboot your appliance. Rebooting your
Asterisk Appliance 50 will terminate any active calls.
If the registration process fails, please confirm that you have entered the
G.729 license key correctly, and that a firewall is not blocking the
Asterisk Appliance 50 from communicating with Digium’s registration
server on TCP port 443.
If you do not currently own a G.729 codec license, click Buy new G729
Codec Licenses to be directed to a page where you can purchase G.729
codec licenses.
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Backup
This is a housekeeping tab which allows you to back up your Asterisk
Appliance 50 configuration to preserve your changes. To create a backup,
click Create New Backup, specify a file name (e.g. the backup date), and
select whether you want to also backup voicemails and custom prompts.
You can then download a previously created backup, restore from the
backup, delete the backup file, or upload a backup from another machine.
Figure 49: Backup Page
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Update
The Updates tab provides an interface for downloading or uploading
newer Asterisk Appliance 50 firmware images, and for downloading
newer Polycom firmware and bootrom images to the Asterisk Appliance
50. Customers with an active Service Subscription may visit the
Digium.com website to register their Asterisk Appliance 50, activate their
Service Subscription, and download software updates for the Asterisk
Appliance 50.
Figure 50: Asterisk Appliance 50 Update
There are two interfaces for putting a new Asterisk Appliance 50
firmware image on the Asterisk Appliance 50. The first section provides
the user the ability to specify a location from which the Asterisk
Appliance 50 will connect and download the updated software. The user
may specify an absolute HTTP location such as: http://company.com/
downloads/software.img, or the address and filename on an accessible
TFTP server. The second section provides a web-based interface for
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uploading software updates. Here, the user may click Browse, select a
local copy of the new software, and click Upload.
Figure 51: Polycom Update
Click Update Polycom firmware to update the Polycom firmware or
bootrom images on the Asterisk Appliance 50. A link is provided under
the Download new Firmware button to get the latest archive names. The
firmware and bootrom archive names must be specified exactly as they
are at that link in order for the update to complete successfully. Click
Download new Firmware after you have specified the latest archive
names in the firmware and bootrom fields.
The download process may take a few minutes to complete, and even
longer if your network bandwidth is limited. A message box will appear
once the firmware has finished downloading.
The Polycom firmware and bootrom images will be installed during the
next reboot cycle of the Asterisk Appliance 50.
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Click Options on the left menu, select the Reboot tab, and then click
Reboot Now to reboot your appliance. Rebooting your Asterisk
Appliance 50 will terminate any active calls.
Note: The next reboot cycle will be increased by approximately 5
minutes during the installation process. In addition, you must reboot
your Polycom phones in order for them to download the new firmware
and bootrom images from the Asterisk Appliance 50.
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Options
The options tab provides several options which allow you to change the
password for your AsteriskGUI logon, modify local extension and agent
settings, as well as reboot the Asterisk Appliance 50. The Advanced tab
allows you to enable or disable advanced options. The basic options are
displayed by default. Please refer to the Advanced Options section for a
description of the advanced options.
Figure 52: Asterisk Appliance 50 Options
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General Preferences
The General Preferences tab gives you several useful global settings for
your Asterisk Appliance 50.
 Global Outbound CID - This setting specifies the default global Cal­
lerID that is used for all outgoing calls when no other CallerID of a
higher priority is specified.
 Global Outbound CID Name - This setting specifies the default Cal­
lerID name that is used for all outgoing calls. You may wish to set this
to your company’s name. Leave this value blank if you want the user’s
CallerID name to appear on outbound calls.
 Operator Extension - Select the user extension from the drop-down
list which will be dialed when a user or caller presses “0” to exit voicemail. It is also available as a Voice Menu option.
 Ring Timeout - Specify the number of seconds to ring a device before
sending a call to a user’s voicemail box or Follow Me numbers.
 Enable Idle Image Display - Select this option to enable the display
of an image on a phones LCD display when the phone is idle.
 VoIP Phone Digit Map - This option gives you the ability to define a
global digit mapping string compatible with RFC 3435, section 2.1.5,
to be used with VoIP phones provisioned by this system. There is no
default setting, and this option does not sync with the dialplan
assigned to an individual user. The following examples should assist in
writing an acceptable digit mapping string.
– [2-9]11 - Calls beginning with digits 2-9, followed by digits 11, are
dialed immediately.
– 0T - Calls beginning with digit 0 are dialed after the Digit Timeout
is reached.
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– +011xxx.T - Calls beginning with the + character, followed by 011


digits, and then at least three more digits before any arbitrary num­
ber is matched, are dialed after the Digit Timeout is reached.
– 0[2-9]xxxxxxxxx - Calls beginning with 0, followed by any digit
from 2-9, followed by 9 more digits, are dialed immediately.
– +1[2-9]xxxxxxxx - Calls beginning with the + character, followed
by 1, followed by any digit from 2-9, followed by 8 more digits, are
dialed immediately.
– [2-9]xxxxxxxxx - Calls beginning with any digit from 2-9, followed
by 9 more digits, are dialed immediately.
– [2-9]xxxT - Calls beginning with any digit from 2-9, followed by
three more digits, are dialed after the Digit Timeout is reached.
– These examples would be represented in this option entry box as:
[2-9]11|0T|+011xxx.T|0[2-9]xxxxxxxxx|+1[2-9]xxxxxxxx|[2­
9]xxxxxxxxx|[2-9]xxxT
Each entry is separated by the | character. For more information,
please refer to RFC 3435.
VoIP Phone Digit Timeout - This option specifies the number of sec­
onds the phone will wait after a digit is dialed before trying to estab­
lish a call when a digit map pattern has not been matched. This must
be defined as an integer.
Extension Preferences - This section gives you the ability to define
the numerical range for all extension types.
Language
The Language tab gives you the ability to specify the default language
for all prompts for phone to phone, inbound, and outbound calls. If a
soundpack selection is made, but the soundpack is not already installed,
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the soundpack will be downloaded from the Digium website. English,
Spanish, and French prompts are loaded by default.
Change Password
The Change Password tab gives you the ability to change your
administrator password.
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Factory Reset
The Factory Reset tab gives you the ability to reset your Asterisk
Appliance 50 to the factory defaults. Warning: If you reset your Asterisk
Appliance 50 to factory defaults, you will lose all configuration changes.
Be sure to make a backup of your configuration before resetting your
Asterisk Appliance 50.
Reboot
The Reboot tab gives you the ability to reboot your Asterisk Appliance
50. Some configuration changes you make may require a system reboot.
Warning: Rebooting the appliance will terminate all active calls.
Advanced Options
There are several advanced options which can be made accessible from
the Options page. This gives advanced users with a background in
Asterisk the ability to refine the Asterisk Appliance 50 configuration.
Clicking Show Advanced Options provides additional advanced options
under certain existing menu pages, and also activates several advanced
menu items on the left hand sidebar.
Note: Digium does not provide support for the options which are
made accessible by selecting Show Advanced Options nor for bugs
discovered in the options which are made accessible by selecting
Show Advanced Options. If your unit becomes inoperable due to the
editing of an option which is made accessible by selecting Show
Advanced Options, Digium Technical Support will request that you
reset your unit to Factory Default configuration.
Note: Any changes made on the advanced options pages must be
activated by clicking Apply Changes at the top of the GUI.
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The following is a list of the advanced menu items which will be made
accessible from the left hand sidebar after selecting Show Advanced
Options:
 Active Channels - This page displays the active channels on the PBX,
along with options to transfer a call to a user on the system or to hang
up a call.
 Bulk Add - This page gives you the ability to add multiple users to the
system in one easy step -- import from a CSV (comma-separated val­
ues) file or create a range of extensions.
 File Editor - This page lets you edit any Asterisk configuration file
within the GUI, as well as create a new configuration file.
 Asterisk CLI - The Asterisk CLI is a command line interface which
can be used for issuing any Asterisk command or series of commands.
The results of the commands are displayed in the pane below the com­
mand line field. Enter Help in the command line field for a list of com­
mands.
 SIP Settings - The advanced SIP settings can be configured from this
page.
 IAX Settings - The advanced IAX settings can be configured from
this page.
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Troubleshooting
Where can I find answers to additional questions?
There are several places to inquire for more information:
1. Digium Technical Support (+1.256.428.6161), or Toll Free in the U.S.
(1.877.344.4861), is available 7am-8pm Central Time (GMT -6),
Monday - Friday.
2. Asterisk users mailing list (asterisk.org/lists.digium.com).
3. IRC channel #asterisk on (irc.freenode.net).
Subscription Services Program
Digium is dedicated to supporting your Asterisk system by offering full
technical support through our Subscription Services Program. The
Subscription service covers the Asterisk Appliance 50. Through this
program, you can be at ease knowing that your business will always have
access to the Asterisk experts. Pricing on Subscription Services may be
obtained from your nearest reseller or you may call Digium Sales for
referral to your nearest reseller at +1.256.428.6000. You can also visit
http://www.digium.com/sales.
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Appendix A
Pin Assignments
This appendix provides pin assignments for the Asterisk Appliance
50 unit.
LAN
Ports
Craft
Port
WAN
Port
Analog
Ports
Figure A-1: Back Panel Ports
All eight analog ports on the Asterisk Appliance 50 are 8-pin RJ11 ports.
The pin assignments are identified in Table A-2.
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Table A-1: CRAFT Port Pinout
Diagram
Pin 1
Pin 8
Pin
Description
1
Ground (Connect to DB9 pin 5)
2
Unused (Leave Open)
3
Primary RxD (To AA50)
(Connect to DB9 Pin 3)
4
Open
5
Tx (From AA50)
(Connect to DB 9 Pin 2)
6
CTS (To AA50)
(Connect to DB9 Pin 7)
7
Open
8
RTS (From AA50) (Connect to
DB9 Pin 8)
The CRAFT port serial parameters are 57600 8N1 (57600 bits per second,
8 data bits, no parity, 1 stop bit), with hardware flow control = no,
software flow control = yes.
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Appendix A: Pin Assignments
Table A-2: RJ11 Analog Port Connector
Diagram
Pin 1
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Pin 6
Pin
Description
1
Unused
2
Unused
3
Tip
4
Ring
5
Unused
6
Unused
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Appendix A: Pin Assignments
Table A-3: LAN & WAN Ethernet Port Pinouts
Diagram
Pin 1
Digium, Inc.
Pin 8
Pin
Description
1
Rx Receive Negative
2
Rx Receive Positive
3
Tx Transmit Negative
4
Unused
5
Unused
6
Tx Transmit Positive
7
Unused
8
Unused
Page 144
Appendix B
Specifications
This appendix provides specifications, required environmental
conditions, and maximum power consumption for the Asterisk
Appliance 50 unit.
Physical.
Size:
Weight:
11.7” × 7.5” × 1.72” (29.72 x 19.05 x 4.37 cm)
Full Assembly 1.4 lbs (635g)
Interfaces.
LAN Ports - Quad RJ45 10/100baseT
WAN Port - RJ 45 10/100baseT
Analog Ports - Octal RJ11
Craft Port - RJ45, 57600 8N1
DC Power - 6.3mm O.D., 2mm pin; 12V 3A center positive
CompactFlash - Type 1
Environment.
Temperature: 0 to 40° C (32 to 104° F) operation
-20 to 70° C (4 to 158° F) storage
Humidity: Up to 90% non-condensing
Digium, Inc.
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Appendix B: Specifications
Table B-4: Maximum 12V Power Consumption
Item
Total
Each FXS port in use with 3REN load
Power
36 Watts
1.5 Watts
Note: Power consumption is determined by the number of analog
phones connected to the FXS ports and the REN rating of the
phones.
Digium, Inc.
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Appendix C
Asterisk Appliance 50 (AA50) License Agreement
DIGIUM END-USER PURCHASE AND LICENSE AGREEMENT
FOR THE ASTERISK APPLIANCE 50
December, 2008
IMPORTANT – PLEASE READ CAREFULLY
1. This Digium End-User Purchase and License Agreement (the
"Agreement") is a legal agreement between Digium and the licensee,
purchaser and end user respectively (hereinafter, “you”, “You” or
“your”) of the Digium distribution media, embedded software and
related documentation (the "Software"), Digium services (“Services”),
and any Digium computer electronics (“Hardware”), entitlements
granted pursuant to a Subscription Agreement, and related manuals
(collectively the "Products"). By downloading or installing the
Software or installing the Hardware, you agree to and accept the terms
and conditions of this Agreement. If you do not accept, or are not
authorized to accept the terms and conditions of this Agreement, then
you should not install the Software or Hardware and should remove
any installed Software and Hardware from your computer.
2. GRANT OF LICENSE. Subject to the terms and conditions of this
Agreement and Exhibit B, Digium grants you a non-exclusive, non­
sublicenseable, non-transferable license to use the Software for
internal business purposes and not for resale, sublicense, leasing, or
providing hosted services to third-parties. “Software” shall include
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
any upgrades, updates, bug fixes or modified versions (“Upgrades”) or
backup copies of the Software supplied to you by Digium or an
authorized reseller, provided you hold a valid license to the original
Software and have paid any applicable fee for Upgrades.
Notwithstanding the foregoing, Digium acknowledges that certain
components of the Software are licensed for distribution under socalled “open source” software licenses (“Open Source Components”).
Digium provides a list of these Open Source Components for the
initially distributed version of the Software in Exhibit B, along with
their respective licenses. To the extent the terms of the license
applicable to an Open Source Component prohibit any of the
restrictions in this Agreement with respect to such Open Source
Component, such restrictions will not apply to such Open Source
Component.
3. RESERVATION OF RIGHTS. Except for the limited license rights
expressly granted in this Agreement, Digium reserves all rights in and
to the Software and any modifications thereto, including title,
ownership, trademark, copyright, patent, trade secret and any other
rights and interests. You will own only the Hardware (exclusive of
Software embedded in the Hardware) and the physical media on which
the Software and associated documentation are reproduced and
distributed. If you allow an existing Services or Subscription
Agreement plan to lapse, you may be required to pay retroactively
annual fees for all Products from the date of the lapse in order to
reinstate such Services. Products that are provided or sold as demo or
evaluation units, or are marked with the legend “For Evaluation Only”
or a similar notation, are not authorized for commercial business use in
production or deployment, but are made available only for
demonstration or evaluation purposes, and may not be resold or
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
transferred to any third party without prior written permission from
Digium.
4. EMERGENCY CALLS. You understand and acknowledge that the
Products may be used to implement, supplement, or replace telephone
systems and telecommunications services, and that in some cases,
certain government regulations may apply to their implementation or
use; and compliance with such regulations is your sole responsibility.
You understand and acknowledge that users of the system on which
you install the Products may attempt to use that system to place
emergency calls. You acknowledge and agree that: the Products must
be properly configured for your system or application; that the nature
of the Products and any networks they may operate upon allow many
possible configurations; that such configuration may be beyond the
scope of the documentation supplied with the Products; and that
specialized experience and training may be required to properly
configure the Products. You acknowledge and agree that it is your sole
responsibility to ensure that the Products and associated networks and
systems are implemented and configured such that emergency calls are
properly handled, and that any system or application based on the
Products complies with all applicable laws and regulations. You
acknowledge and agree that telephone and telecommunications
systems can be complex and must be installed, implemented, and
configured by the appropriate technically qualified personnel, and that
you or your authorized agents have the qualifications necessary to
properly implement and configure the Products to handle emergency
calls, if applicable. You further acknowledge and agree that it is your
sole and ongoing responsibility to ensure the proper operation of any
emergency calling system based on the Products, including, but not
limited to: initially and regularly testing the operation of the Products,
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
including testing the operation with emergency services; notifying
and training all users of any system on which the Products are installed
how to use the system for emergency calls; and notifying such users of
any and all limitations of your configuration and implementations of
the Products and any network or system the Products are used on or
with. By using the Products under this Agreement, you explicitly
release Digium from any warranty, duty, liability, or obligation to train
you or any users of your system regarding the proper configuration,
operation, or use of the Products or any system or network they are
used in conjunction with on which it is installed; to ensure that your
configuration, implementation, or use of the Products provides for the
proper handling or routing of emergency calls; or to ensure that your
use of the Products is in compliance with any applicable laws and
regulations.
5. LIMITED WARRANTY
5.1 Limited Warranty - Software. Digium warrants that the
Software will substantially conform to Digium’s specifications as set
forth in the applicable documentation, at and from the date of original
retail purchase of the Software for a period of ninety days provided
that the Software is properly installed and used as contemplated in its
documentation. Digium’s sole obligation and your sole remedy for a
breach of the foregoing warranty shall be to replace the non­
conforming Software with software substantially conforming to
Digium’s specifications or to refund fees as described below. Digium
further warrants that the media on which Digium delivers the Software
will be free of physical defects for a period of ninety days or will be
replaced by Digium. Except as otherwise agreed by Digium in
writing, the replacement Software is subject to the terms and
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
conditions of this Agreement. If Digium determines in its sole
discretion that a material non-conformance cannot be corrected, or that
it is not practical to replace the nonconforming Software, the price
paid by the original licensee for the Software will be refunded by
Digium to you and the license grant automatically terminated, in
which case you will promptly return to Digium or destroy the
Software and any copies in your possession.
5.2 Limited Warranty - Hardware. Digium warrants that the
Hardware will be free from material defects in workmanship and
materials from the date of original purchase by you for a period of two
years. Digium’s sole obligation shall be to repair or replace the
defective Hardware during the warranty period at no charge to you or
to refund as described below. Such repair or replacement will be
performed by Digium or its authorized representative. Digium may in
its sole discretion replace the defective Hardware (or any part thereof)
with a reconditioned product determined by Digium as materially
equivalent or better than the defective Hardware. Repaired or
replacement Hardware continues tolling the original warranty period
from the date of original retail purchase.
5.3 Limited Warranty – Services AND SUBSCRIPTION
AGREEMENTS. Digium warrants it will perform Services and/or
deliver entitlements (which shall be defined in a Subscription
Agreement) to you in a manner consistent with reasonably applicable
industry standards. As your sole remedy and Digium’s sole liability
for breach of the foregoing warranty, Digium will re-perform or
remedy at no charge to you any non-conforming performance reported
by you within ten days of Digium's receipt of notification of breach of
the foregoing warranty. Digium shall include technical support and/or
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
deliver the Subscription Agreement entitlements for the Products in
accordance with the policies, terms and conditions set forth on the
Digium web site for the Products you purchase. You may elect to
renew your support or Subscription Agreement from Digium at
Digium’s then current rates unless such Products and associated
Services are discontinued in Digium’s sole discretion. You will be
informed of the subscription entitlements, and terms and conditions
(Subscription Agreement) during the Product registration process. A
current copy of the entitlements may be viewed on Digium's website
and is incorporated in this Agreement by reference. Digium may
amend the Subscription Agreement at any time. You are responsible
for reviewing Digium's website for changes in the terms and
conditions of the Subscription Agreement. In the case where Product
registration is required in order to activate a Subscription Agreement,
that Subscription Agreement shall be considered activated and legally
binding beginning upon the date of Product registration. However,
should you fail to register the product within thirty (30) days of your
receipt of Product, the Subscription Agreement shall be considered
activated and legally binding beginning on the thirty first (31st) day
following your receipt of the Product.
5.4 Warranty Disclaimers. The limited warranties provided by
Digium do not cover (i) Products that have been subjected to misuse,
tampering, modification, experimentation, alteration, negligence,
faulty installation; (ii) Products with the model or serial number
altered, tampered with, or removed; (iii) Initial installation, installation
and removal of the Product for repair, and shipping costs; (iv)
Configuration of the Product; (v) Products that are provided or sold as
demo or evaluation units, or are marked with the legend “For
Evaluation Only” or a similar notation; (vi) Damage that occurs in
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
shipment due to act of God, failures due to power surge, and cosmetic
damage; (vii) Any hardware, software, firmware or other materials or
services provided by anyone other than Digium; or (viii) Repair by
anyone other than Digium or Digium’s authorized representative.
Unless under a current Services contract with Digium, you shall not
upgrade or update Product using software of any origin except
commercial versions of the Software released to you by Digium or its
authorized distributors from time to time. the product is provided
without any implied warranties of any kind whatsoever, including,
without limitation, any implied warranty of merchantability, fitness for
a particular purpose and non-infringement. Except as expressly
covered under the limited warranty provided herein, digium disclaims
all laibility as to the quality, selection and performance.
6. Return Material Authorization. If returning a Product under
warranty, you must include with the Product a written description of
the Hardware defect or Software nonconformance in sufficient detail
to allow Digium to confirm such determination. Prior to returning a
Product under warranty to Digium, the original purchaser must obtain
a Return Material Authorization (“RMA”) in accordance with the
Digium policy in effect at the time as published on the Digium web
site. You must package the warranted Product in the original shipping
package to ensure that it will not be damaged in transit, with the RMA
number prominently marked on the outside of the package. No
manuals or accessories should be returned to Digium, and Digium will
only refund or replace the defective portion of the Product, but will not
ship back any other material. Digium will ship repaired or replaced
Products to you via UPS Ground or any common carrier selected by
Digium, with shipping charges prepaid. You may request and prepay
expedited shipping. Digium has no responsibility for packages that
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
are lost in transit to Digium. Digium will not be obligated to accept
physical return of any Software.
7. LIMITATION OF LIABILITY. To the maximum extent permitted
by law, Digium is not liable under any contract, negligence, strict
liability or other legal or equitable theory for any loss of use of the
Products, inconvenience or indirect damages of any character, whether
special, incidental or consequential (including, but not limited to, loss
of revenue or profit, work stoppage, computer failure or malfunction,
failure of connected equipment or programs, loss of information or
data or loss of goodwill) resulting from the use of the Products,
relating to warranty service, or arising out of any breach of this
Agreement, even if Digium has been advised of the possibility of such
damages. The sole remedy for a breach of the foregoing limited
warranty is repair, replacement or refund of the defective or non­
conforming product. The maximum liability of Digium under this
Agreement is limited to the purchase price of the applicable Products.
The foregoing express written warranties and remedies are exclusive
and in lieu of any other warranties or remedies, express, implied or
statutory.
8. PROPRIETARY WORKS. The Products contain trademarks, trade
secrets and/or copyrighted materials of Digium or its suppliers.
8.1 You agree not to reverse engineer, decompile, or disassemble the
Software, except to the extent such restriction is expressly prohibited
by applicable law. You shall not disclose or make available such trade
secrets or copyrighted material in any form to any third party nor
remove any trademark notices, copyright notices, or licensing terms
from the Software or any components therein.
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
8.2 You will not, without Digium written consent, use the name,
trademarks, trade names or logos of Digium, or the name of any
product or service of Digium, in any manner. If Digium grants you a
right to use the aforementioned, you will do so only in strict
compliance with Digium trademark policies.
8.3 You agree not to fork any version of the components of Software
licensed to you by Digium or promote or contribute to any fork of the
components of Software licensed to you by Digium. For purposes of
this Agreement, "fork" means a royalty free GNU General Purpose
License (“GPL”) or GPL-compatible distribution of the Software that
is independent of the Digium-maintained Software code base.
9. TERMINATION. This Agreement shall terminate upon either
destruction of the Products or return of the Products by you to Digium.
In the event of a breach of the scope of use permitted by the grant in
Section 2, or if you do not comply with other materials terms and
conditions of this Agreement, Digium shall have the right to
immediately terminate this Agreement, in which case you must
promptly destroy or return all Products to Digium. Notwithstanding
the foregoing, the provisions of Sections 5.4, 7, 8, 9, 10, 11, 12, 13 and
14 shall survive termination of this Agreement.
10. EXPORT RESTRICTION. You acknowledge that the Software,
with the possible exception of certain third-party components, is of
U.S. origin. You agree to comply with applicable domestic and
international laws that apply to the Software, including the U.S.
Export Administration Regulations and other limitations issued by
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
U.S. and foreign governments.
11. TRANSFER AND ASSIGNMENT. This Agreement and the rights
and obligations under it are not assignable by you without the prior
written approval of Digium, voluntarily or by operation of law. Any
attempt by you to assign this Agreement without such approval shall
be void. This Agreement shall inure to the benefit of the successors
and assigns of Digium. Notwithstanding the foregoing, you may move
the Software to different internal computers to the extent consistent
with the scope of license you have purchased to the Software.
12. U.S. GOVERNMENT USERS. The Software and documentation
qualify as “commercial items” as defined at 48 C.F.R. 2.101 and 48
C.F.R. 12.212. All Government users acquire the Software and
documentation with only those rights herein that apply to non­
governmental customers of Digium.
13. GOVERNING LAW AND JURISDICTION. This Agreement is to
be construed in accordance with and governed by laws of the State of
Alabama, excluding its conflict of law provisions. Digium and you
agree to submit to the personal and exclusive jurisdiction of, and agree
that venue is proper in, the Alabama State or Federal Courts located in
the County of Madison, Alabama, for any such legal action or
proceeding. The United Nations Convention on International Sale of
Goods, the application of which is expressly excluded, does not
govern this Agreement.
14. ENTIRE AGREEMENT. This Agreement and Exhibit B constitute
the entire understanding between the parties relating to the subject
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
matter hereof and supersede all prior writings, negotiations or
understandings with respect thereto. The provisions of this Agreement
shall take precedence over any conflicting terms in any subsequent
purchase order, documentation or collateral. The parties agree that
this Agreement may be executed electronically and that electronic
copies of this Agreement shall be binding upon the parties.
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
EXHIBIT B – OPEN SOURCE COMPONENTS
1. OPEN SOURCE LICENSES. Digium acknowledges that the
components of Software listed in this Exhibit are licensed for
distribution under so-called “open source” distribution licenses, and
includes the licenses for those components in compliance with the
license terms.
You may obtain distribution media containing the source code for
these components from Digium by contacting Digium at:
Postal Service:
Digium, Inc.
Sales Department
445 Jan Davis Drive NW
Huntsville, AL 35806
Electronic mail:
http://www.digium.com/sales
Via either means, you may order product code 8AA50GPLCD, which
has a unit cost of US$1.00, and will be shipped to you after payment of
shipping handling charges (US$20.00 for United States destinations
and US$50.00 for all other destinations). Orders placed via electronic
mail must be paid via a Digium-accepted credit card; orders via postal
service can be paid either via credit card or a check drawn on a United
States bank.
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
2. GNU GENERAL PUBLIC LICENSE VERSION 2
COMPONENTS. The following components are licensed under
version 2 of the GNU General Public License (included as Exhibit
B.1).
Linux® kernel
libnetold
ext2fs
uuid
libipq
libiptc
busybox
gdbserver
e2fsck
mke2fs
iwconfig
inetd
ssmtp
dnsmasq
dhcpcd
version
expand
Linux is a registered trademark of Linus Torvalds.
3. GNU LESSER GENERAL PUBLIC LICENSE VERSION 2
COMPONENTS. The following components are licensed under
version 2 of the GNU Lesser General Public License (included as
Exhibit B.9).
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
uClibc
gettext
libiconv
gmime
glib
4. OTHER OPEN SOURCE LICENSES. The following components
are licensed under various licenses, included in the listed exhibits.
Digium, Inc.
ncurses
Exhibit B.2
dropbear
Exhibit B.3
OpenSSL
Exhibit B.4
ftp
Exhibit B.5
ntp and ntpdate
Exhibit B.6
zlib
Exhibit B.7
ss and et libraries
Exhibit B.8
libpcap
Exhibit B.10
tcpdump
Exhibit B.11
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
EXHIBIT B.1 - GNU General Public License Version 2
GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
Copyright (C) 1989, 1991 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Everyone is permitted to copy and distribute verbatim copies of this
license document, but changing it is not allowed.
Preamble
The licenses for most software are designed to take away your freedom to
share and change it. By contrast, the GNU General Public License is
intended to guarantee your freedom to share and change free software--to
make sure the software is free for all its users. This General Public
License applies to most of the Free Software Foundation's software and to
any other program whose authors commit to using it. (Some other Free
Software Foundation software is covered by the GNU Library General
Public License instead.) You can apply it to your programs, too.
When we speak of free software, we are referring to freedom, not price.
Our General Public Licenses are designed to make sure that you have the
freedom to distribute copies of free software (and charge for this service if
you wish), that you receive source code or can get it if you want it, that
you can change the software or use pieces of it in new free programs; and
that you know you can do these things.
To protect your rights, we need to make restrictions that forbid anyone to
deny you these rights or to ask you to surrender the rights. These
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
restrictions translate to certain responsibilities for you if you distribute
copies of the software, or if you modify it.
For example, if you distribute copies of such a program, whethergratis or
for a fee, you must give the recipients all the rights that you have. You
must make sure that they, too, receive or can get the source code. And
you must show them these terms so they know their rights.
We protect your rights with two steps: (1) copyright the software, and (2)
offer you this license which gives you legal permission to copy, distribute
and/or modify the software.
Also, for each author's protection and ours, we want to make certain that
everyone understands that there is no warranty for this free software. If
the software is modified by someone else and passed on, we want its
recipients to know that what they have is not the original, so that any
problems introduced by others will not reflect on the original authors'
reputations.
Finally, any free program is threatened constantly by software patents.
We wish to avoid the danger that redistributors of a free program will
individually obtain patent licenses, in effect making the program
proprietary. To prevent this, we have made it clear that any patent must
be licensed for everyone's free use or not licensed at all.
The precise terms and conditions for copying, distribution and
modification follow.
GNU GENERAL PUBLIC LICENSE
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains a
notice placed by the copyright holder saying it may be distributed
under the terms of this General Public License. The "Program",
below, refers to any such program or work, and a "work based on the
Program" means either the Program or any derivative work under
copyright law: that is to say, a work containing the Program or a
portion of it, either verbatim or with modifications and/or translated
into another language. (Hereinafter, translation is included without
limitation in the term "modification".) Each licensee is addressed as
"you".
Activities other than copying, distribution and modification are not
covered by this License; they are outside its scope. The act of running the
Program is not restricted, and the output from the Program is covered
only if its contents constitute a work based on the Program (independent
of having been made by running the Program). Whether that is true
depends on what the Program does.
1. You may copy and distribute verbatim copies of the Program's source
code as you receive it, in any medium, provided that you
conspicuously and appropriately publish on each copy an appropriate
copyright notice and disclaimer of warranty; keep intact all the notices
that refer to this License and to the absence of any warranty; and give
any other recipients of the Program a copy of this License along with
the Program.
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
You may charge a fee for the physical act of transferring a copy, and you
may at your option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion of
it, thus forming a work based on the Program, and copy and distribute
such modifications or work under the terms of Section 1 above,
provided that you also meet all of these conditions:
a) You must cause the modified files to carry prominent notices stating
that you changed the files and the date of any change.
b) You must cause any work that you distribute or publish, that in whole
or in part contains or is derived from the Program or any part thereof, to
be licensed as a whole at no charge to all third parties under the terms of
this License.
c) If the modified program normally reads commands interactively when
run, you must cause it, when started running for such interactive use in the
most ordinary way, to print or display an announcement including an
appropriate copyright notice and a notice that there is no warranty (or
else, saying that you provide a warranty) and that users may redistribute
the program under these conditions, and telling the user how to view a
copy of this License. (Exception: if the Program itself is interactive but
does not normally print such an announcement, your work based on the
Program is not required to print an announcement.)
These requirements apply to the modified work as a whole. If identifiable
sections of that work are not derived from the Program, and can be
reasonably considered independent and separate works in themselves,
then this License, and its terms, do not apply to those sections when you
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
distribute them as separate works. But when you distribute the same
sections as part of a whole which is a work based on the Program, the
distribution of the whole must be on the terms of this License, whose
permissions for other licensees extend to the entire whole, and thus to
each and every part regardless of who wrote it.
Thus, it is not the intent of this section to claim rights or contest your
rights to work written entirely by you; rather, the intent is to exercise the
right to control the distribution of derivative or collective works based on
the Program.
In addition, mere aggregation of another work not based on the Program
with the Program (or with a work based on the Program) on a volume of a
storage or distribution medium does not bring the other work under the
scope of this License.
3. You may copy and distribute the Program (or a work based on it, under
Section 2) in object code or executable form under the terms of
Sections 1 and 2 above provided that you also do one of the following:
a) Accompany it with the complete corresponding machine-readable
source code, which must be distributed under the terms of Sections 1 and
2 above on a medium customarily used for software interchange; or,
b) Accompany it with a written offer, valid for at least three years, to give
any third party, for a charge no more than your cost of physically
performing source distribution, a complete machine-readable copy of the
corresponding source code, to be distributed under the terms of Sections 1
and 2 above on a medium customarily used for software interchange; or,
Digium, Inc.
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Appendix C: Asterisk Appliance 50 (AA50) License Agreement
c) Accompany it with the information you received as to the offer to
distribute corresponding source code. (This alternative is allowed only
for noncommercial distribution and only if you received the program in
object code or executable form with such an offer, in accord with
Subsection b above.)
The source code for a work means the preferred form of the work for
making modifications to it. For an executable work, complete source
code means all the source code for all modules it contains, plus any
associated interface definition files, plus the scripts used to control
compilation and installation of the executable. However, as a special
exception, the source code distributed need not include anything that is
normally distributed (in either source or binary form) with the major
components (compiler, kernel, and so on) of the operating system on
which the executable runs, unless that component itself accompanies the
executable.
If distribution of executable or object code is made by offering access to
copy from a designated place, then offering equivalent access to copy the
source code from the same place counts as distribution of the source code,
even though third parties are not compelled to copy the source along with
the object code.
4. You may not copy, modify, sublicense, or distribute the Program
except as expressly provided under this License. Any attempt
otherwise to copy, modify, sublicense or distribute the Program is
void, and will automatically terminate your rights under this License.
However, parties who have received copies, or rights, from you under
this License will not have their licenses terminated so long as such
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parties remain in full compliance.
5. You are not required to accept this License, since you have not signed
it. However, nothing else grants you permission to modify or
distribute the Program or its derivative works. These actions are
prohibited by law if you do not accept this License. Therefore, by
modifying or distributing the Program (or any work based on the
Program), you indicate your acceptance of this License to do so, and
all its terms and conditions for copying, distributing or modifying the
Program or works based on it.
6. Each time you redistribute the Program (or any work based on the
Program), the recipient automatically receives a license from the
original licensor to copy, distribute or modify the Program subject to
these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein. You
are not responsible for enforcing compliance by third parties to this
License.
7. If, as a consequence of a court judgment or allegation of patent
infringement or for any other reason (not limited to patent issues),
conditions are imposed on you (whether by court order, agreement or
otherwise) that contradict the conditions of this License, they do not
excuse you from the conditions of this License. If you cannot
distribute so as to satisfy simultaneously your obligations under this
License and any other pertinent obligations, then as a consequence you
may not distribute the Program at all. For example, if a patent license
would not permit royalty-free redistribution of the Program by all
those who receive copies directly or indirectly through you, then the
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only way you could satisfy both it and this License would be to refrain
entirely from distribution of the Program.
If any portion of this section is held invalid or unenforceable under any
particular circumstance, the balance of the section is intended to apply
and the section as a whole is intended to apply in other circumstances.
It is not the purpose of this section to induce you to infringe any patents or
other property right claims or to contest validity of any such claims; this
section has the sole purpose of protecting the integrity of the free software
distribution system, which is implemented by public license practices.
Many people have made generous contributions to the wide range of
software distributed through that system in reliance on consistent
application of that system; it is up to the author/donor to decide if he or
she is willing to distribute software through any other system and a
licensee cannot impose that choice.
This section is intended to make thoroughly clear what is believed to be a
consequence of the rest of this License.
8. If the distribution and/or use of the Program is restricted in certain
countries either by patents or by copyrighted interfaces, the original
copyright holder who places the Program under this License may add
an explicit geographical distribution limitation excluding those
countries, so that distribution is permitted only in or among countries
not thus excluded. In such case, this License incorporates the
limitation as if written in the body of this License.
9. The Free Software Foundation may publish revised and/or new
versions of the General Public License from time to time. Such new
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versions will be similar in spirit to the present version, but may differ
in detail to address new problems or concerns.
Each version is given a distinguishing version number. If the Program
specifies a version number of this License which applies to it and "any
later version", you have the option of following the terms and conditions
either of that version or of any later version published by the Free
Software Foundation. If the Program does not specify a version number
of this License, you may choose any version ever published by the Free
Software Foundation.
10. If you wish to incorporate parts of the Program into other free
programs whose distribution conditions are different, write to the
author to ask for permission. For software which is copyrighted by the
Free Software Foundation, write to the Free Software Foundation; we
sometimes make exceptions for this. Our decision will be guided by
the two goals of preserving the free status of all derivatives of our free
software and of promoting the sharing and reuse of software generally.
NO WARRANTY
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE,
THERE IS NO WARRANTY FOR THE PROGRAM, TO THE
EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
OTHERWISE STATED IN WRITING THE COPYRIGHT
HOLDERS AND/OR OTHER PARTIES PROVIDE THE
PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND,
EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR
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PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND
PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD
THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST
OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR
AGREED TO IN WRITING WILL ANY COPYRIGHT HOLDER,
OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE
LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL
DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE
THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
DATA OR DATA BEING RENDERED INACCURATE OR LOSSES
SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF
THE PROGRAM TO OPERATE WITH ANY OTHER
PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS
BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Programs
If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it free
software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the program. It is safest to attach
them to the start of each source file to most effectively convey the
exclusion of warranty; and each file should have at least the "copyright"
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line and a pointer to where the full notice is found.
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) 19yy <name of author>
This program is free software; you can redistribute it and/or modify it
under the terms of the GNU General Public License as published by the
Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful, but
WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software Foundation,
Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this when it
starts in an interactive mode:
Gnomovision version 69, Copyright (C) 19yy name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details
type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
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The hypothetical commands `show w' and `show c' should show the
appropriate parts of the General Public License. Of course, the
commands you use may be called something other than `show w' and
`show c'; they could even be mouse-clicks or menu items--whatever suits
your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James
Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications
with the library. If this is what you want to do, use the GNU Library
General Public License instead of this License.
Digium, Inc.
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EXHIBIT B.2 - ncurses Licenses
Top-level license:
------------------------------------------------------------------------------- Copyright (c) 1998-2004,2006 Free Software Foundation, Inc.
-­
---- Permission is hereby granted, free of charge, to any person obtaining a
­
-
-- copy of this software and associated documentation files (the
-­
-- "Software"), to deal in the Software without restriction, including
­
-
-- without limitation the rights to use, copy, modify, merge, publish,
­
-
-- distribute, distribute with modifications, sublicense, and/or sell copies ­
-
-- of the Software, and to permit persons to whom the Software is furnished ­
-
-- to do so, subject to the following conditions:
-­
---- The above copyright notice and this permission notice shall be included
­
-
-- in all copies or substantial portions of the Software.
-­
---- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
­
-
-- OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
-­
-- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN ­
-
-- NO EVENT SHALL THE ABOVE COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
-­
-- DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
­
-
-- OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE ­
-
-- USE OR OTHER DEALINGS IN THE SOFTWARE.
-­
---- Except as contained in this notice, the name(s) of the above copyright
­
-
-- holders shall not be used in advertising or otherwise to promote the
-­
-- sale, use or other dealings in this Software without prior written
-­
-- authorization.
--
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libmenu and libform:
/****************************************************************************
* Copyright (c) 1998-2003,2004 Free Software Foundation, Inc.
*
*
*
* Permission is hereby granted, free of charge, to any person obtaining a *
* copy of this software and associated documentation files (the
*
* "Software"), to deal in the Software without restriction, including
*
* without limitation the rights to use, copy, modify, merge, publish,
*
* distribute, distribute with modifications, sublicense, and/or sell
*
* copies of the Software, and to permit persons to whom the Software is
*
* furnished to do so, subject to the following conditions:
*
*
*
* The above copyright notice and this permission notice shall be included *
* in all copies or substantial portions of the Software.
*
*
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS *
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
*
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
*
* IN NO EVENT SHALL THE ABOVE COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
*
* DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
*
* OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR
*
* THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*
*
* Except as contained in this notice, the name(s) of the above copyright
*
* holders shall not be used in advertising or otherwise to promote the
*
* sale, use or other dealings in this Software without prior written
*
* authorization.
*
****************************************************************************/
Digium, Inc.
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libpanel:
/****************************************************************************
* Copyright (c) 1998-2004,2005 Free Software Foundation, Inc.
*
*
*
* Permission is hereby granted, free of charge, to any person obtaining a *
* copy of this software and associated documentation files (the
*
* "Software"), to deal in the Software without restriction, including
*
* without limitation the rights to use, copy, modify, merge, publish,
*
* distribute, distribute with modifications, sublicense, and/or sell
*
* copies of the Software, and to permit persons to whom the Software is
*
* furnished to do so, subject to the following conditions:
*
*
*
* The above copyright notice and this permission notice shall be included *
* in all copies or substantial portions of the Software.
*
*
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS *
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
*
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
*
* IN NO EVENT SHALL THE ABOVE COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
*
* DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
*
* OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR
*
* THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*
*
*
* Except as contained in this notice, the name(s) of the above copyright
*
* holders shall not be used in advertising or otherwise to promote the
*
* sale, use or other dealings in this Software without prior written
*
* authorization.
*
****************************************************************************/
Digium, Inc.
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EXHIBIT B.3 - dropbear License
Dropbear contains a number of components from different sources, hence
there are a few licenses and authors involved. All licenses are fairly non­
restrictive.
The majority of code is written by Matt Johnston, under the license below.
Portions of the client-mode work are (c) 2004 Mihnea Stoenescu, under
the same license:
Copyright (c) 2002-2004 Matt Johnston
Portions copyright (c) 2004 Mihnea Stoenescu
All rights reserved.
Permission is hereby granted, free of charge, to any person obtaining a
copy of this software and associated documentation files (the "Software"),
to deal in the Software without restriction, including without limitation
the rights to use, copy, modify, merge, publish, distribute, sublicense, and/
or sell copies of the Software, and to permit persons to whom the
Software is furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in
all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY
OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT
LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND
NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
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COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES
OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR
IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
=====
LibTomCrypt and LibTomMath are written by Tom St Denis, and are
Public Domain.
=====
sshpty.c is taken from OpenSSH 3.5p1,
Copyright (c) 1995 Tatu Ylonen <ylo@cs.hut.fi>, Espoo, Finland
All rights reserved
"As far as I am concerned, the code I have written for this software
can be used freely for any purpose. Any derived versions of this
software must be clearly marked as such, and if the derived work is
incompatible with the protocol description in the RFC file, it must be
called by a name other than "ssh" or "Secure Shell". "
=====
loginrec.c
loginrec.h
atomicio.h
atomicio.c
and strlcat() (included in util.c) are from OpenSSH 3.6.1p2, and are
licensed
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under the 2 point BSD license.
loginrec is written primarily by Andre Lucas, atomicio.c by Theo de
Raadt.
strlcat() is (c) Todd C. Miller
=====
Import code in keyimport.c is modified from PuTTY's import.c, licensed
as
follows:
PuTTY is copyright 1997-2003 Simon Tatham.
Portions copyright Robert de Bath, Joris van Rantwijk, Delian Delchev,
Andreas Schultz, Jeroen Massar, Wez Furlong, Nicolas Barry, Justin
Bradford, and CORE SDI S.A.
Permission is hereby granted, free of charge, to any person obtaining a
copy of this software and associated documentation files (the "Software"),
to deal in the Software without restriction, including without limitation
the rights to use, copy, modify, merge, publish, distribute, sublicense, and/
or sell copies of the Software, and to permit persons to whom the
Software is furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in
all copies or substantial portions of the Software.
Digium, Inc.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY
OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT
LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND
NONINFRINGEMENT. IN NO EVENT SHALL THE COPYRIGHT
HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
THE SOFTWARE.
Digium, Inc.
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EXHIBIT B.4 - OpenSSL License
The OpenSSL toolkit stays under a dual license, i.e. both the conditions of
the OpenSSL License and the original SSLeay license apply to the toolkit.
See below for the actual license texts. Actually both licenses are BSDstyle Open Source licenses. In case of any license issues related to
OpenSSLplease contact openssl-core@openssl.org.
OpenSSL License
--------------/* ====================================================================
* Copyright (c) 1998-2006 The OpenSSL Project. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
*
notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
*
notice, this list of conditions and the following disclaimer in
*
the documentation and/or other materials provided with the
*
distribution.
*
* 3. All advertising materials mentioning features or use of this
*
software must display the following acknowledgment:
*
"This product includes software developed by the OpenSSL Project
*
for use in the OpenSSL Toolkit. (http://www.openssl.org/)"
*
* 4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to
*
endorse or promote products derived from this software without
*
prior written permission. For written permission, please contact
*
openssl-core@openssl.org.
*
* 5. Products derived from this software may not be called "OpenSSL"
*
nor may "OpenSSL" appear in their names without prior written
*
permission of the OpenSSL Project.
*
* 6. Redistributions of any form whatsoever must retain the following
*
acknowledgment:
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*
"This product includes software developed by the OpenSSL Project
*
for use in the OpenSSL Toolkit (http://www.openssl.org/)"
*
* THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND ANY
* EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE OpenSSL PROJECT OR
* ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
* STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
* OF THE POSSIBILITY OF SUCH DAMAGE.
* ====================================================================
*
* This product includes cryptographic software written by Eric Young
* (eay@cryptsoft.com). This product includes software written by Tim
* Hudson (tjh@cryptsoft.com).
*
*/
Original SSLeay License
----------------------/* Copyright (C) 1995-1998 Eric Young (eay@cryptsoft.com)
* All rights reserved.
*
* This package is an SSL implementation written
* by Eric Young (eay@cryptsoft.com).
* The implementation was written so as to conform with Netscapes SSL.
* * This library is free for commercial and non-commercial use as long as
* the following conditions are aheared to. The following conditions
* apply to all code found in this distribution, be it the RC4, RSA,
* lhash, DES, etc., code; not just the SSL code. The SSL documentation
* included with this distribution is covered by the same copyright terms
* except that the holder is Tim Hudson (tjh@cryptsoft.com).
* * Copyright remains Eric Young's, and as such any Copyright notices in
* the code are not to be removed.
* If this package is used in a product, Eric Young should be given
attribution
* as the author of the parts of the library used.
* This can be in the form of a textual message at program startup or
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* in documentation (online or textual) provided with the package.
* * Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the copyright
*
notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
*
notice, this list of conditions and the following disclaimer in the
*
documentation and/or other materials provided with the distribution.
* 3. All advertising materials mentioning features or use of this software
*
must display the following acknowledgement:
*
"This product includes cryptographic software written by
*
Eric Young (eay@cryptsoft.com)"
*
The word 'cryptographic' can be left out if the rouines from the library
*
being used are not cryptographic related :-).
* 4. If you include any Windows specific code (or a derivative thereof) from
*
the apps directory (application code) you must include an
acknowledgement:
*
"This product includes software written by Tim Hudson
(tjh@cryptsoft.com)"
* * THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
* * The licence and distribution terms for any publically available version or
* derivative of this code cannot be changed. i.e. this code cannot simply be
* copied and put under another distribution licence
* [including the GNU Public Licence.]
*/
Digium, Inc.
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EXHIBIT B.5 - ftp License
/*
* Copyright (c) 1985, 1989 Regents of the University of California.
* All rights reserved.
* * Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
*
notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
*
notice, this list of conditions and the following disclaimer in the
*
documentation and/or other materials provided with the distribution.
* 3. All advertising materials mentioning features or use of this software
*
must display the following acknowledgement:
* This product includes software developed by the University of
* California, Berkeley and its contributors.
* 4. Neither the name of the University nor the names of its contributors
*
may be used to endorse or promote products derived from this software
*
without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
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EXHIBIT B.6 - ntp and ntpdate License
***********************************************************************
*
*
* Copyright (c) David L. Mills 1992-2001
*
*
*
* Permission to use, copy, modify, and distribute this software and
*
* its documentation for any purpose and without fee is hereby
*
* granted, provided that the above copyright notice appears in all
*
* copies and that both the copyright notice and this permission
*
* notice appear in supporting documentation, and that the name
*
* University of Delaware not be used in advertising or publicity
*
* pertaining to distribution of the software without specific,
*
* written prior permission. The University of Delaware makes no
*
* representations about the suitability this software for any
*
* purpose. It is provided "as is" without express or implied
*
* warranty.
*
*
*
***********************************************************************
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EXHIBIT B.7 - zlib License
Copyright (C) 1995-2002 Jean-loup Gailly and Mark Adler
This software is provided 'as-is', without any express or implied warranty.
In no event will the authors be held liable for any damages arising from
the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it freely,
subject to the following restrictions:
1. The origin of this software must not be misrepresented ; you must not
claim that you wrote the original software. If you use this software in a
product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not
be misrepresented as being the original software.
3. This notice may not be removed or altered from any source
distribution.
Jean-loup Gailly
jloup@gzip.org
Mark Adler
madler@alumni.caltech.edu
Digium, Inc.
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EXHIBIT B.8 - ss and et License
/*
* Copyright 1987, 1988, 1989 Massachusetts Institute of Technology
* (Student Information Processing Board)
*
* Permission to use, copy, modify, and distribute this software and
* its documentation for any purpose is hereby granted, provided that
* the names of M.I.T. and the M.I.T. S.I.P.B. not be used in
* advertising or publicity pertaining to distribution of the software
* without specific, written prior permission. M.I.T. and the
* M.I.T. S.I.P.B. make no representations about the suitability of
* this software for any purpose. It is provided "as is" without
* express or implied warranty.
*/
Digium, Inc.
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EXHIBIT B.9 – GNU Lesser General Public License Version 2
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies of this
license document, but changing it is not allowed.
[This is the first released version of the Lesser GPL. It also counts as the
successor of the GNU Library Public License, version 2, hence the
version number 2.1.]
Preamble
The licenses for most software are designed to take away your freedom to
share and change it. By contrast, the GNU General Public Licenses are
intended to guarantee your freedom to share and change free software--to
make sure the software is free for all its users.
This license, the Lesser General Public License, applies to some specially
designated software packages--typically libraries--of the Free Software
Foundation and other authors who decide to use it. You can use it too, but
we suggest you first think carefully about whether this license or the
ordinary General Public License is the better strategy to use in any
particular case, based on the explanations below.
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When we speak of free software, we are referring to freedom of use, not
price. Our General Public Licenses are designed to make sure that you
have the freedom to distribute copies of free software (and charge for this
service if you wish); that you receive source code or can get it if you want
it; that you can change the software and use pieces of it in new free
programs; and that you are informed that you can do these things.
To protect your rights, we need to make restrictions that forbid
distributors to deny you these rights or to ask you to surrender these
rights. These restrictions translate to certain responsibilities for you if
you distribute copies of the library or if you modify it.
For example, if you distribute copies of the library, whether gratis or for a
fee, you must give the recipients all the rights that we gave you. You
must make sure that they, too, receive or can get the source code. If you
link other code with the library, you must provide complete object files to
the recipients, so that they can relink them with the library after making
changes to the library and recompiling it. And you must show them these
terms so they know their rights.
We protect your rights with a two-step method: (1) we copyright the
library, and (2) we offer you this license, which gives you legal
permission to copy, distribute and/or modify the library.
To protect each distributor, we want to make it very clear that there is no
warranty for the free library. Also, if the library is modified by someone
else and passed on, the recipients should know that what they have is not
the original version, so that the original author's reputation will not be
affected by problems that might be introduced by others.
Digium, Inc.
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Finally, software patents pose a constant threat to the existence of any free
program. We wish to make sure that a company cannot effectively restrict
the users of a free program by obtaining a restrictive license from a patent
holder. Therefore, we insist that any patent license obtained for a version
of the library must be consistent with the full freedom of use specified in
this license.
Most GNU software, including some libraries, is covered by the ordinary
GNU General Public License. This license, the GNU Lesser General
Public License, applies to certain designated libraries, and is quite
different from the ordinary General Public License. We use this license
for certain libraries in order to permit linking those libraries into non-free
programs.
When a program is linked with a library, whether statically or using a
shared library, the combination of the two is legally speaking a combined
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Ty Coon, President of Vice
That's all there is to it!
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EXHIBIT B.10 – libpcap License
Copyright (c) 1993, 1994, 1995, 1996, 1997, 1998
The Regents of the University of California. All rights reserved.
Redistribution and use in source and binary forms, with or without
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must display the following acknowledgement:
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Engineering Group at Lawrence Berkeley Laboratory.
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specific prior written permission.
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THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED
WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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EXHIBIT B.11 – tcpdump License
License: BSD
Redistribution and use in source and binary forms, with or without
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Note: Copyrights referred to in the above license are too numerous to
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individual copyright declarations and the modules to which they apply.
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Appendix D
Glossary and Acronyms
ANSI
American National Standards Institute
An organization which proposes and establishes standards for
international communications.
asynchronous
Not synchronized; not timed to an outside clock source. Transmission is
controlled by start bits at the beginning and stop bits at the end of each
character. Asynchronous communications are often found in internet
access and remote office applications.
attenuation
The dissipation of a transmitted signal’s power as it travels over a wire.
bandwidth
The capacity to carry traffic. Higher bandwidth indicates the ability to
transfer more data in a given time period.
bit
The smallest element of information in a digital system. A bit can be
either a zero or a one.
bps
bits per second
A measurement of transmission speed across a data connection.
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Appendix D: Glossary and Acronyms
broadband
Broadband transmission shares the bandwidth of a particular medium
(copper or fiber optic) to integrate multiple signals. The channels take up
different frequencies on the cable, integrating voice, data, and video over
one line.
channel
A generic term for an individual data stream. Service providers can use
multiplexing techniques to transmit multiple channels over a common
medium.
Cat5
Category of Performance for wiring and cabling. Cat 5 cabling support
applications up to 100 MHz.
Cat5E
Category of Performance for wiring and cabling. Category 5 Enhanced
wiring supports signal rates up to 100 MHz but adheres to stricter quality
specifications.
CLEC
competitive local exchange carrier
A term for telephone companies established after the
Telecommunications Act of 1996 deregulated the LECs. CLECs compete
with ILECs to offer local service. See also LEC and ILEC.
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Appendix D: Glossary and Acronyms
CO
central office
The CO houses local switching equipment. All local access lines in a
particular geographic area terminate at this facility (which is usually
owned and operated by an ILEC).
CPE
customer premises equipment
Terminal equipment which is connected to the telecommunications
network and which resides within the home or office of the customer. This
includes telephones, modems, terminals, routers, and television set-top
boxes.
DS0
Digital Signal, Level 0
A voice grade channel of 64 Kbps. The worldwide standard speed for
digitizing voice conversation using PCM (Pulse Code Modulation).
DS1
Digital Signal, Level 1
1.544 Mbps in North America (T1) and Japan (J1) -up to 24 voice
channels (DS0s), 2.048 Mbps in Europe (E1) - up to 32 voice channels
(DS0s). DS1/T1/E1 lines are part of the PSTN.
DS3
Digital Signal, Level 3
T3 in North America and Japan, E3 in Europe. Up to 672 voice channels
(DS0s). DS3/T3/E3 lines are not part of the PSTN
DTMF
Dual Tone Multi-Frequency
Push-button or touch tone dialing.
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Appendix D: Glossary and Acronyms
E1
The European equivalent of North American T1, transmits data at 2.048
Mbps, up to 32 voice channels (DS0s).
E3
The European equivalent of North American T3, transmits data at 34.368
Mbps, up to 512 voice channels (DS0s). Equivalent to 16 E1 lines.
EMI
Electromagnetic Interference
Unwanted electrical noise present on a power line
full duplex
Data transmission in two directions simultaneously.
FXO
Foreign Exchange Office
Receives the ringing voltage from an FXS device. Outside lines are
connected to the FXO port on your Asterisk Appliance 50 unit.
FXS
Foreign Exchange Station
Initiates and sends ringing voltage. Phones are connected to the FXS ports
on the Asterisk Appliance 50 unit.
G.711
A recommendation by the Telecommunication Standardization Sector
(ITU-T) for an algorithm designed to transmit and receive mulaw PCM
voice and A-law at a digital bit rate of 64 Kbps. This algorithm is used for
digital telephone sets on digital PBX.
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Appendix D: Glossary and Acronyms
G.723.1
A recommendation by the Telecommunication Standardization Sector
(ITU-T) for an algorithm designed to transmit and receive audio over
telephone lines at 6.3 Kbps or 5.3 Kbps.
G.729a
A recommendation by the Telecommunication Standardization Sector
(ITU-T) for an algorithm designed to transmit and receive audio over
telephone lines at 8 Kbps.
H.323
A recommendation by the Telecommunication Standardization Sector
(ITU-T) for multimedia communications over packet-based networks.
IAX
Inter-Asterisk eXchange
A VoIP protocol used by Asterisk. It is used to enable VoIP connections
between Asterisk servers, and between servers and clients that also use
the IAX protocol.
iLBC
internet Low Bitrate Codec
A free speech codec used for voice over IP. It is designed for narrow band
speech with a payload bitrate of 13.33 kbps (frame length = 30ms) and
15.2 kbps (frame length = 20 ms).
ILEC
incumbent local exchange carrier
The LECs that were the original carriers in the market prior to the entry of
competition and therefore have the dominant position in the market.
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Appendix D: Glossary and Acronyms
interface
A point of contact between two systems, networks, or devices.
ISO
International Standards Organization
LED
light-emitting diode
Linux
A robust, feature-packed open source operating system based on Unix
that remains freely available on the internet. It boasts dependability and
offers a wide range of compatibility with hardware and software. Asterisk
is supported exclusively on Linux.
loopback
A state in which the transmit signal is reversed back as the receive signal,
typically by a far end network element.
MGCP
Media Gateway Control Protocol
multiplexing
Transmitting multiple signals over a single line or channel. FDM
(frequency division multiplexing) and TDM (time division multiplexing)
are the two most common methods. FDM separates signals by dividing
the data onto different carrier frequencies, and TDM separates signals by
interleaving bits one after the other.
MUX
multiplexer
A device which transmits multiple signals over a single communications
line or channel. See multiplexing.
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Appendix D: Glossary and Acronyms
PBX
private branch exchange
A smaller version of a phone company’s large central switching office.
Example: Asterisk.
PCI
peripheral component interconnect
A standard bus used in most computers to connect peripheral devices.
POP
point of presence
The physical connection point between a network and a telephone
network. A POP is usually a network node serving as the equivalent of a
CO to a network service provider or an interexchange carrier.
POTS
plain old telephone service
Standard phone service over the public switched telephone network
(PSTN). This service provides analog bandwidth of less than 4 kHz.
PPP
point-to-point protocol
Type of communications link that connects a single device to another
single device, such as a remote terminal to a host computer.
PSTN
public switched telephone network
The public switched telephone network (PSTN) is the network of the
world's public circuit-switched telephone networks. Originally a network
of fixed-line analog telephone systems, the PSTN is now almost entirely
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Appendix D: Glossary and Acronyms
digital, and now includes mobile as well as fixed telephones.
QoS
quality of service
A measure of telephone service, as specified by the Public Service
Commission.
RJ11
A six-pin jack typically used for connecting telephones, modems, and fax
machines in residential and business settings to PBX or the local
telephone CO.
SIP
Session Initiation Protocol
An IETF standard for setting up sessions between one or more clients. It
is currently the leading signaling protocol for Voice over IP, gradually
replacing H.323.
T1
A dedicated digital carrier facility which transmits up to 24 voice
channels (DS0s) and transmits data at 1.544 Mbps. Commonly used to
carry traffic to and from private business networks and ISPs.
T3
A dedicated digital carrier facility which consists of 28 T1 lines and
transmits data at 44.736 Mbps. Equivalent to 672 voice channels (DS0s).
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Appendix D: Glossary and Acronyms
TDM
time division multiplexer
A device that supports simultaneous transmission of multiple data streams
into a single high-speed data stream. TDM separates signals by
interleaving bits one after the other.
telco
A generic name which refers to the telephone companies throughout the
world, including RBOCs, LECs, and PTTs.
tip and ring
The standard termination on the two conductors of a telephone circuit;
named after the physical appearance of the contact areas on the jack plug.
twisted pair
Two copper wires commonly used for telephony and data
communications. The wires are wrapped loosely around each other to
minimize radio frequency interference or interference from other pairs in
the same bundle.
V
volts
VoIP
Voice over IP
Technology used for transmitting voice traffic over a data network using
the Internet Protocol.
Zaptel (Zap)
Zapata Telephony Project dedicated to implementing a reasonable and
affordable Computer Telephony platform into the world marketplace.
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