Digium AA50 Administrator's Manual

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Asterisk Appliance™ 50

(AA50)

Administrator Manual

601-00005 Rev. C

445 Jan Davis Drive

Digium, Inc.

Huntsvil le, 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

No part of this publication may be copied, distributed, transmitted, transcribed, stored in a retri eval syst em , or t ran sl ated int o any huma n or compu te r lan gu ag e with ou t t he pr io r wri tte n permission of Digium, Inc.

Digium, Inc. has made every effort to ensure that the instructions contained in this document are ade q u a te an d error fr ee . The ma nu f a ctu r e r w i ll, if nec es s ar y , ex plain is su es w h ic h m ay not be covered by this documentation. The manufacturer’s liability for any errors in the docume nt is limited to the correction of errors and the aforementioned advisor y services.

This doc ument has been prepar ed for us e by profe ssiona l and pr operly tr ained personn el, and the cus to m er as su m es full respon si bi li t y whe n us ing it.

Adobe and Acrobat are registered trademarks, and Acrobat Reader is a trademark of Adobe Systems Incorporated.

Asteri sk and D igi um a r e re gi ster e d tr ad emar ks and Ast eri sk B usi ne ss Ed it i on, A st eri sk NOW, AsteriskGUI, an d Asterisk Appliance 50 are trademarks of D igium, Inc.

Any other trademarks mentioned in the document are the property of their respective owners.

Digium, Inc. Page 2

Safety Certificat ion and Agency Approvals Safety:

US/CSA 60950 IEC 60950 EN 60950 AS/NZ 60950

Other:

CE Mark (European Union) 2002/95/EC Restr ictions on Hazardous S ubst ances (Ro HS), 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 enclosur e 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 teleph one company.

A plug and jack used to connect this equipment to the premi ses 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 har m 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 a ffect 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 repa ir and/or war ranty inf ormatio n. 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 interferen ce, and (2) T h is dev ice mu s t accep t any in terference received , including interf erence 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

1 2 3

A B C

Title Description

Overview Identifies the features of your unit.

Unit Installation Provides instructions for installing the unit.

Asterisk Configuration

Troublesh ooting Explain s resolutions to common problems and

Pin Assignments Lists the connectors and pin assignments .

Specifications Details unit specifications.

License Agreeme nt Digium End-User Purchase and Li cens e Agreement

Glossary and Acronyms

Provide s instructions on how to configure the Embedded Asterisk Business Edition through the use of the AsteriskGUI.

frequentl y as ked questions pertaining to the unit.

Defines terms related to this product.

Digium, Inc. Page 6

Symbol Definitions

Caution stat emen ts in dicate a c onditio n whe r e d amage to t he un it o r 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 sym b o l in d i ca t es electrostat i c sen si ti ve device s. O b serve prec autions for handling devi ces. Wear a proper ly grounded electrostatic discha rge (ESD) wrist strap while handling the device.

The Electrical Hazard Symbol indicates a possibility of electrical shock when operat ing this unit in certain situations. To reduce the risk of damage or injury, fol low all steps or proc edures as instructed.

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Important Safety Instructions

Servicing.

Do not attempt to servi ce this unit un less s pecif ic ally ins truc ted to do so. Do not attempt to remove the unit from your equipment while power is present. Refe r ser vicing to qualified service personnel.

Water and Moisture. Do not spill liquids on this unit. Do not operate this equipment in a wet environme nt.

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 anythi ng (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 enclosur e or gain acc es s to areas where you ar e 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 91 1 or Emergency calls are placed shou ld ens ur e the ir pr o vid er pr o perl y forw ar ds t he cus tomer 's accessibl e 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 Applia n ce 50 Docume nt a tion . . . . . . . . . . .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 Asteris k Appl iance 50 Interface . . . . . . . . . . . . . . . . . . . . . . . .32

Analog Hardware Conf i gura tion . . . . . . . . . . . . . . . . . . . . . . . . . . .35

Trunk Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .40

Analog Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .41

Outgoing Ca lling Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .55

Dial Plans . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .59

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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

Incomi ng Ca lling Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .89

Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .93

Paging/Intercom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97

Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 01

Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .104

Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .110

Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 12

Voicemail Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .122

System Info . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .123

Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .124

G.729 Codec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 27

Backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 30

Update . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 31

Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 34

Chapter 4

Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 40

Appendix A

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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: The Asteris k Appl iance 50 (AA50) . . . . . . . . . . . . . .17

Figure 2: Example Asterisk Appliance 50 Port Identification . .22

Figure 3 : Asterisk Appliance 50 Back View . . . . . . . . . . . . . . .24

Figure 4 : Mounting Instructions . . . . . . . . . . . . . . . . . . . . . . . .27

Figure 5: GUI Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

Figure 6 : System Status Page . . . . . . . . . . . . . . . . . . . . . . . . .32

Figure 7 : Configure Hardware . . . . . . . . . . . . . . . . . . . . . . . . .35

Figure 8 : Trunk Configuration Page . . . . . . . . . . . . . . . . . . . . .40

Figure 9 : New Analog Trunk Definition . . . . . . . . . . . . . . . . . . 41

Figure 10: Edit Analog Trunk Definition . . . . . . . . . . . . . . . . . . .45

Figure 1 1: Add New Service Provider . . . . . . . . . . . . . . . . . . . .47

Figure 1 2: Edit VoIP Service Provider . . . . . . . . . . . . . . . . . . . . 48

Figure 1 3: Create New SIP/IAX Trunk Definition . . . . . . . . . . . .50

Figure 14: Ed it VoIP Trunk . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52

Figure 15: Outbound Calling Rules . . . . . . . . . . . . . . . . . . . . . .55

Figure 16: New Calling Rule . . . . . . . . . . . . . . . . . . . . . . . . . . .56

Figure 1 7: Create New Dial Plan . . . . . . . . . . . . . . . . . . . . . . . .59

Figure 18: Use r Exte n sions . . . . . . . . . . . . . . . . . . . . . . . . . . . .61

Figure 19: Cre a te New User . . . . . . . . . . . . . . . . . . . . . . . . . . .62

Figure 2 0: New Ring Group . . . . . . . . . . . . . . . . . . . . . . . . . . . .68

Figure 21: Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .70

Figure 22: New Call Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72

Figure 2 3: Default Voice Menu . . . . . . . . . . . . . . . . . . . . . . . . .78

Figure 2 4: Custom Voice Menu Prompts Page . . . . . . . . . . . . .85

Figure 2 5: Record Menu Prompts . . . . . . . . . . . . . . . . . . . . . . .85

Figure 26: Up load Menu Prompts . . . . . . . . . . . . . . . . . . . . . . .86

Figure 2 7: New Time Interval . . . . . . . . . . . . . . . . . . . . . . . . . . .87

Figure 28: Incoming Calling Rules . . . . . . . . . . . . . . . . . . . . . . .89

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List of Figures

Figure 29: Incoming Calling Rules . . . . . . . . . . . . . . . . . . . . . . .90

Figure 3 0: Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .93

Figure 3 1: Paging/Intercom . . . . . . . . . . . . . . . . . . . . . . . . . . . .97

Figure 32: New Page/Intercom Group . . . . . . . . . . . . . . . . . . . .98

Figure 3 3: Settings for Paging Individual Extensions . . . . . . . . .99

Figure 3 4: Paging & Intercom Settings . . . . . . . . . . . . . . . . . . 1 00

Figure 3 5: New Conference Bridge . . . . . . . . . . . . . . . . . . . . .101

Figure 3 6: Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 04

Figure 37: New Follow Me Definition . . . . . . . . . . . . . . . . . . . .106

Figure 3 8: Follow Me Options . . . . . . . . . . . . . . . . . . . . . . . . .108

Figure 3 9: Directory Settings . . . . . . . . . . . . . . . . . . . . . . . . . .110

Figure 4 0: Feature Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . .112

Figure 4 1: Call Parking Preferences . . . . . . . . . . . . . . . . . . . .116

Figure 4 2: Application Map . . . . . . . . . . . . . . . . . . . . . . . . . . .1 18

Figure 4 3: Dial Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .120

Figure 4 4: New Voicemail Group . . . . . . . . . . . . . . . . . . . . . . .122

Figure 4 5: System Information . . . . . . . . . . . . . . . . . . . . . . . . .1 23

Figure 4 6: Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .124

Figure 4 7: G.729 Codec Registration . . . . . . . . . . . . . . . . . . .127

Figure 4 8: G.729 Codec License Information . . . . . . . . . . . . .1 28

Figure 4 9: Backup Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . .130

Figure 5 0: Asterisk Appliance 50 Update . . . . . . . . . . . . . . . . .1 31

Figure 5 1: Polycom Update . . . . . . . . . . . . . . . . . . . . . . . . . . .132

Figure 5 2: Asterisk Appliance 50 Options . . . . . . . . . . . . . . . . 1 34

Figure A-1: Back Panel Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 141

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List of Tabl e s

Table A-1: CRAFT Port Pinout . . . . . . . . . . . . . . . . . . . . . . . . .142

Table A-2: RJ11 Analog Port Connector. . . . . . . . . . . . . . . . . .1 43

Table A-3: LAN & WAN Ethernet Port Pinouts . . . . . . . . . . . . .144

Table B-4: Maximum 12V Power Consumption . . . . . . . . . . . . 1 46

Digium, Inc. 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 funct ion 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 configur ed 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 in terface f or the Asteris k Applia nce 50. I t gives you the ability to conf igure the basic har dware and dia l plan elemen ts 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

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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 10/100Base T 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 , S80 8i with Eight FXO, and S844i with Four FXS and Four FXO)

 SIP and IAX2 VoIP protocols  CompactFlash interfa ce (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 R evision B) or 128ms (Hardware Revision C) of ana-

log port echo cancellation

 8MB on board serial Flash memory  64MB 16 bit parallel SDRAM  Fron t panel LE Ds

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  Identifyi ng C om m un ic a tion Por ts on page 19  Underst andi n g th e LEDs on page 19  Using the Configuration Reset Switch on page 23  Installin g the A st eris k Appliance 50 on page 24  Mounting the Asterisk Appliance 50 on page 27

Figure 1: The Asterisk Appliance 50 (AA50)

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Chapter 2: Unit Installation

Unpacking the Unit

When you unpack your unit, carefully inspect it for any damage that m ay have occurred during shipmen t. If damage is suspected, file a claim with the carrier and contact your resell er from which the unit was purcha sed or Digium T ec hnical Support (+1. 256.428. 6161). Keep the origina l shi pping 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 thi s function themselves.

Inspecting Your Shipment

The following items are includ ed in shipment of the Asterisk Appliance 50:

 Asterisk Appliance 50 (AA50)  Compact Fl ash Card  Power Supply  Power Cable  Analog Cables (optional depending on model)  CD-ROM containing manual and installa tion files  Product Registration Card  Support and Warranty Information

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Chapter 2: Unit Installation

Identifyi ng C om m un ic a tion Por ts

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 por t echo canc ellation. 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 th e ports and their cor responding LEDs .

Underst andi n g th e 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.

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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 c omplete d. The LE D pulses at a rate which is proporti onal 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.

LAN (4 ports)

Green (flashing)

Orange (flashing)

Off No line is connected or the interface is

Green (flashing)

Orange (flashing)

Link is up at 100Mbps. LED flashes at 1/ 10 second intervals as traffic is detected.

Link is up at 10Mbps. LED flashes at 1/10 second intervals as traffic is detected.

inactive. Link is up at 100Mbps. LED flashes at 1/

10 second intervals as traffic is detected. Link is up at 10Mbps. LED flashes at 1/10

second intervals as traffic is detected.

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Chapter 2: Unit Installation

Table 1: LED Definitions

LED Color Description

Analog (8 ports)

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

Telephone is ringing.

(flashing) Green (slow

Telephone is in use.

blinking) Red (solid) Port is configured for FXO operation and

is enabled. A telephone line may be connected to this port.

Red

T elephone line is ringing.

(flashing) Red (slow

T elephone line is in use.

blinking)

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Chapter 2: Unit Installation

Figure 2: Example Asterisk Appliance 50 Port Identification

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Chapter 2: Unit Installation

Using the Configuration Reset Switch

The Configuration Reset (CFG RST) switch (rear panel) will reset the current Asterisk Appli ance 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 de lete all configuration data. The administrator password will also be reset. See Figure 3 on page 24 to locate the

switch.

RST

Caution.

Pressin g th e CFG RST sw itch will ca u s e lo ss o f al l configuration settings and reset administration passwords .

CFG

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Chapter 2: Unit Installation

Configuration Reset Sw itch

Craft

Port

WAN

Port

Power

Supply

Analog

Ports

LAN

Ports

Installin g the A st eris k Appliance 50

1. Remove the Compact Flash cover plate and insert the Compact Flash

Figure 3: Asterisk Appliance 50 Back View

card before connecting the power supply.

Caution.

The Compact Flash is not hot swappable. The Compact Flash card sh ould be inserted before powering on the unit. Likewise, before removing the Compact Flash card it should be unmounted (using the unmount command) and the Asteri s k Appliance 50 should be powered off.

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Chapter 2: Unit Installation

2. Connect one end of an Ethernet cable to an Asterisk Appliance 50

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 Aster isk Appliance 50 to your computer so that you may access the Asterisk Appliance 50 GUI from your computer.

3. 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 Ast eris k Appl iance 50 will immedia tely power on once connected to a power source.

4. Using an Asterisk Appliance 50 supporte d web browser, open a

browser window and enter the IP address for the Asterisk Appliance

50. The default LAN I P address i s 192 .168.69. 1. The def ault use rname 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 ne w password. Once th e log on process is complete the AsteriskGUI home page will be displaye d.

5. You may find it preferable t o enable the Aster isk App liance 5 0 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.

6. Select the Enable GUI on WAN interface checkbox.

7. Click Save, and then click Apply Changes. Your changes will be

applied and Asterisk will reload.

8. Attach the eth ernet cabl e co nn ect ed to t he Asteri sk App lia nce 5 0 ’s

LAN port to the WAN port. Connect the other end of the cable to the appropriate inte rnet connection (will vary depending on your setup). This will connect the Asteri sk Appliance 50 to the internet.

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9. Connect telephones to the analog por ts tha t are configured as FXS

ports and conne ct phon e line s to th e analog por ts that ar e confi gured 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 firs t logging into the Asterisk Appliance 50.

11. You are now ready to configure your Asterisk Applianc e 50 via the

GUI.

Caution.

This unit must be connected to the Teleco mmu nications 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 anythi ng (including paper) on top of th e Asterisk Appliance 50. To allow proper cooling, these units must not be stacked.

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Table 2: W a ll Moun ting

Step Instru c tio n s for Wall Mounting

1 Select the area to mount the Asterisk Appliance 50 unit

(ref er 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-inc h or long er)

into the desired location on the wall . They should be placed approximately 7 1/2-inc he s, or 19cm, apar t 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 th e head of the scr ews to slide i nto 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 point ing downward ma y cause a disruption in air circulation, which could cause the Asterisk Appliance 50 to overheat. Mo unting t he As teri sk Appl ianc e 50 th is way ca n also expose the LAN, WAN, and analog ports to potential damage.

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Chapter 3 Telephone System Configuration

This chapter provides infor mation 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 Asteris k Appliance 50 Inte rface on page 32  Analog Hardware Configuration on page 35  Trunk C onfiguration on page 40  Outgoing Calling Rules on page 55  Dial Plans on page 59  User Extens i ons on page 61  Ring Groups on page 68  Music on Hold on page 70  Call Queu es 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  Voic e m a il 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 int erface. The AsteriskGUI gives you the ability to set up your telephone system withou t the need to use command line configuration. Aft er connecti ng to the Asterisk Appli ance 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 alr eady be connected to an int ernet or network connection, as described in Installing the Asterisk Appliance 50 on page 24. In the address field of an Asterisk Appli ance 50 supported web browser , e nter t he IP addre ss assi gned to your Aste risk Appli ance 50. The default LAN IP address is 192.168.69.1.

Figure 5: GUI Login

T o log on to the system enter the following credenti als:

 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 pr ocess. Once the log on process is complete the AsteriskGUI home page will be displaye d.

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The Asteris k Appliance 50 Inte rface

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 le ft co lumn identif ies a ll the elements for which you can program the Asteri sk Applia nce 50. The elements listed b egin w i th Sys tem Status, which is the first page you see upon logon, and proc eed down to Options. Clicking any of the tabs on the left of the page opens the correspo nding pa ge in the right column. Many pages have additional information. Click on the information symbol, a blue “i” enclosed in a circl e, to get more informati on about a field or page.

The System Status page is the default page. This page shows you the current version of firmwar e 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 definit ions 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 extens ion will have the status of “Unavaila ble” when the VoIP account associated with it is not registered to the Asterisk Appliance 50. The s ta tus will n ot c hange to “Unava ilable” when a use r 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 Configu re Hardw a re 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), e tc . The number of FXS and FXO ports available for configuration will depend on the Asterisk Appliance 50 model you purchased. Click the C onfigure 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 use s a short to r eques t a dial tone . All Nort h American home phone lines use loop sta rt si gnalling. Kewl S tart is t he s ame as Loop S tart , but is better able to detect dis connects. 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 drop- down 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, Ring er Threshold, Current limiting, T ip/ Ring voltage adjustment , Minimum Operational Loop curre nt, and and AC Impedance selection as predefined for each countries analog line characteristic s. Select the country in which your Asterisk Applianc e 50 is operating.

 A-law Override - Set th e 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 Appli ance 50.

 FXS Hono r M ode - 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 volt age to a normal le vel, or Peak will set the volt age

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 experien cin g trouble detecting Caller ID

from Analog service provide rs or whose lines exhibit a polarity rever sal before Calle r ID is tr ansmitted fr om the p rovider s hould sele ct Full Wave. Otherwise, choose Standard.

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 MWI Mode - This option allows the user to specify the type of Mes-

sage W a iti ng Indica tor det ection to be done on tr unk (FXO) i nterfac es. The option s are none, which performs no detecti on, FSK which performs Frequency Shift Key detect ion, or NEON which perform Neon MWI dection. The default value is none.

 Echo Cancellation NLP Type - This option allows you to specif y the

type of Non Linear Processor you want appli ed to the post echo- cancelled audio reflections received from analog connections. There are several options:

– None - This setting disables NLP proces sing 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 con nect ed to the appliance is speak in g . For use rs in quiet environments, Mute may be acceptable.

– Random Noise - This sett ing ca uses t he NLP to i nject r andom no ise

to mask the echo reflecti on. For users in normal env ironments, Ran- dom Noise may be acceptable.

– Hoth N oi se - This setting c auses the NLP to in ject a low-end Gauss-

ian noise with a frequency spectrum similar to voice. For users in normal environments, Hoth Noise may be acceptable.

– Suppressio n N LP - 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 al lows you to spec-

ify the threshold, in dB difference between the received audio (post echo cancellation) and the transm itted 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 N LP Type option is set to Sup- pression NLP, specifies the maximum amount of dB that the NLP should attenu ate 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 ha rdware 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, sel ect the Reboot tab, and then click Reboot Now to reboot your appliance. Rebooting your Asterisk Applia nce 50 will terminate any active calls.

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Trunk C onfiguration

Now that you have c onfigured your analog har dware (assuming your unit had any) you are ready to set up your trunk lines. T runks 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 de finit ions are used i n c alling rules, dial pl ans , and call r outin g, et c. 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 N ame - 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 . Sele ct 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 be fore 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 number 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 milli secon ds of f. Without Busy Pattern specified, the Asterisk Appliance 50 will accept a ny regular sound-silence pattern 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 - T runk (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 recept ion of

a polarity reversal will mark when an outgoing call is answered by the remote party.

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 Hangu p on Po la ri t y Swi tc h - In some countries, a polari t y reversal is

used to signal the disconnect (or hang up) on a phone line . If the Hangup on Polarity Switch option is ena bled, the call will be considered “hung up” on a polarity reversal.

 Call Progress - On trunk interfaces it c an be useful to follow the prog-

ress of a call throug h 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 Zo ne - 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 St art - 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 reversal is detected.

 Caller ID - This opt ion allows the lines to report the c aller ID st ri ng as

received from the telco, or as a fixed val ue by using the adv anced option.

 Pulse Dial - If this option is e nabled, pulse dialing, instead of DTMF,

will be used.

 CID Signalling - This opt ion d efines the typ e of caller I D signalli ng t o

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 messag e waiting indicator received

across the asso ciat ed tr u nk to be fo rw ar de d to a local Use r, such as a SIP phone.

 Flas h Timing - Flash Timing defines the duration, in milliseconds,

that Asterisk will use if it is sen ding a flash signal to another system.

 Receive Flash T iming - Rec eive Flash T im ing define s t he durati on, in

milliseconds, that Asterisk requires in order to consider a flash ope ration it receives to be valid.

Once you have completed the Analog T runk 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 asso ciated 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. Pleas e 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 approximate ly 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 T runk definition, click Update. I n order for these changes to be complet ed, you must reboot your Asterisk Appliance 50. Click Options on the left menu, select the Reboot ta b, and then click Reboot Now to reboot your appliance. Rebooting your Asterisk Appliance 50 will te rminate 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 dyna mically 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 servic e provider it wil l appear in the Ser vice Providers list. There are Edit and Delete buttons assoc iate d with each Servic e Provider listi ng. Click Edit to further refine your service provider defi nition. A detailed definition will be displa yed.

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Figure 12: Edit VoIP Service Provider

The Edit Service Prov ider 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 crede n-

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 fie ld.

 Codecs - Codecs pr ovide the a bility f or you r voic e to b e conv erted t o a

digital signal and transm itted across the Internet. The quality of your call can be affect ed 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 avail able 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 Can cel 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 displ ayed.

Figure 13: Create New SIP/IAX Trunk Definition

Fill in the i nitial 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 rec eives calls using the

VoIP protocol IAX.

 Provider Name - Enter a unique name to help you identify this trunk

for use in calling rules, etc.

 Host name - The hostna me or IP address assigne d to t he VoIP provider

or server.

 Usernam e/P a sswo r d - 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 defini tion, 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.

 Host name - The hostna me or IP address assigne d to t he VoIP provider

or server.

 Username/Password - You will need to provide your log on crede n-

tials in order to update your service provider information.

 Codecs - Codecs pr ovide the a bility f or you r voic e to b e conv erted t o a

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choose. You can select the order in which the codecs are used. The codecs commonly avail able 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 CallerID with this option. Thus this option may be ignored. When making outbound calls the follo wing 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 requir ed by your provider, specify your primary

domain identity to show in the domain fiel d 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 provid er, 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 whet her 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 valu e if you do not want t o re quire a uthentic ation

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. Typical ly a user’s voicemail is stored locally on the Asterisk Appliance 50. The notification of new voice mail is provided by the same lo cal Asterisk Appli ance 50. I f you would like to receive voicemail notifications from a remote provider, this option i s available. To enable this option, clic k the check box, and in the Remote Mail Box field, specify the remote mail box number or identity to whi ch you wi sh t o sub scrib e, e.g. 6001. Sele ct th e loc al use r 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 mai l. It is not possible to provide local voice mail and remote MWI simu ltaneously.

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 pair s an exten sion pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different tr unks (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 m anages onl y indiv idual outg oing call rules. See the Dial Plans section to a ssocia te multipl e 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 initia lly 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 disp layed .

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

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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, spe cify the pattern in this rule similar to the example. All patterns begin with the unders core “_” 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 di git within the br ackets, in this instance 1, 2 , 3, 6,

7, 8, 9.

– . - The period is the wildcard which will match anyth ing 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 destinatio n specif ied 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 check- box is selected.

 Use Trunk - Specify the trunk through whic h calls, matching the spec-

ified pattern, will be placed.

 Strip - This option gives you t he a bility t o r emove spec ified num ber of

digits from the front of the call string before the call is dialed and placed thro ugh the trunk specified in Use Trunk.

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 Prepend Thes e Digi ts - This option gives you the opportunity to add

digits to the front of the call str ing 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 use d to ensure that a call

goes through i f 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 availa ble, then this option will suit your needs.

Once you have completed the calling rul e defi nition 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 ru le, 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 extensi ons to specif y the dialing permissions associated with that extension. For example, you might have one Dial Plan for local calling that only permit s extensions associated with that Dial Plan to dial local numbers, via the "local" outgoing calling rule. Another extension may be permitted to dia l long dista nce numbers, and so wou ld have a Dia l Plan that includes both the "local" and "longdistance" outgoing calling rules.

Click New at the top of the Calling Rules page and c rea te a new dial plan name. You can then add calling rules for that dial plan definition.

Figure 1 7: Cr eate New Di al Pl an

The default dial plan, the colle ction of your calling rules, is Default_Dialplan. You can create more than one dial plan, especially if you want to have di ff erent dial pl an s for different user e xtensio ns. Change the DialPlanName, and then select the checkbox fo r eac h 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 clic k 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 li st to edit a previously defined plan, or Delete to delete a dial plan.

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User Extens i ons

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 n eede d for v oic email, conferenc ing , call que u es, dia l plans, etc. Click the Users tab to view the main User Extensions page.

Figure 18: User Extensions

The main pag e lists al l previ o us ly crea t ed user ex t ens i ons. You can edit individual users as well as cha nge attr ibutes 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

 Extension - The numbered extension, e.g. 6000, assigned to the

defined user. The extension must be a number within the range specified in Extension Preferences on the Options page.

 Name - The first and last name of the indivi dua l 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 rul es you’ve created, you can restrict the outbound dialing of this exte nsion to local calls, emergency calls, and standard long-dis tance calls for North America. This opti on also possibly allows blocking or allowing international (011 prefix dialed) calls.

 Caller ID - Iden tifies the Ca ller ID presented when the listed exten-

sion dials an internal extension.

 Outbound Caller ID - Identifies the Caller ID presented when the

listed extension dia ls an extenal number. Your ability to manipulate your outbound CID may be limited by your VoIP provider. Manipulation 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.

 V oice Mail Access Pin Code - The password used to access voicemai l

for the specified ext ens ion .

 E-mail Address - Voice mails received by this extension can be sent

as audio file attachments e-mailed to a specific address.

 SIP - Identifies wh et her the ex ten s ion s en ds and rece ive s ca lls usin g

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 Asteris k Appliance 50 you may need to confirm the port number liste d on the back of the Aster isk Appliance

50.

 Flash - Flash Timing defines the duration, in milliseconds, that Aster-

isk will use if it is sending a flash signal to a nother system.

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 RXFlash - Receive Flash Time defines the duration, in milliseconds,

that Asterisk requir es 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 service provider you c hoose. You can select the order in which t he codec s 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 P olyCom

® phone connect ed to t he Aste risk Applianc e 50.

The MAC address associates the phone with thi s extension and enables the auto-synchr oniz ation 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 associat ed with this user (e.g. Let’s

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says you configur e a single Polycom phone with two users. User 6000 with Line Number set t o 1 and Line Key se t to 2 will disp lay use r 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 - Tr y 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 prob lems, you usually have problems with your NAT configuration or your firewall's confi guration of SIP and RTP ports.

 Can Reinvite - By default, the Asterisk Appliance 50 will route the

media streams from SIP en dpoints through itself. Enabling this option causes the Ast erisk Appliance 50 to a ttempt 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 routing. This option can be used to tell the Asterisk Appliance 50 whether or not to issue a reinvite to the client.

 DTMF Mod e - Set t he default DTMF mode for sendin g 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 - Insecu re is a SIP paramet er used to determine peer match-

ing. The setting determine s whether or not an insec ure conne ction 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-

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 valu e if you do not want t o re quire a uthentic ation

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 - I f call waiting is not enabled, the extension accepts

only one call before it is identi fied as busy.

 CTI - Selecting this option (Computer Telephony Integration) all ows

the user to connect appli cati o ns to the As te risk Ma na g em en t 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 t he Call Queues opti on.

 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 extension.

Once you have completed the user extension definition click Save to accept the defi niti on , 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 prev iously 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 use rs, o r 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 (ri ng all) or one person can a nswer a call from any phone. The Asterisk Appliance 50 does not come with a default ring group. To create a new ring group click Ne w Ring Gr oup 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 gro up without a ny user extensions.

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T o 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.

Note: Go to Options, Ge ner a l Preferenc es to see which range of numbers have been specified for ri ng gro ups.

3. 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.

4. Choose a ring group strategy from the Strategy drop-down list. You

can choose either Ring All which wil l ring all phones in the defined group simultaneousl y, or Ring Order which will ring phones in sequence determined by the order of the users or trunks in the group.

5. Specify the number of seconds that each phone (or all phones) should

ring before ringing the next phone in order.

6. Lastly, determine which action you want the syste m to take if no one

answers the call. You can either direct the call to the voice ma il 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 accep table, 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 a cceptable format? Linux u sers should tr y the Sox util ity, 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 creat e 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, et c.

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Call Queu es

A call queue lines up callers and allows them to wait to speak to any group of empl oyees taking a high volume of cal ls. The feature allows you to speak to more peopl e rat her t han send call ers back to voice mail to leave a messa ge and receive a call back wh en time permi t s.

Asterisk identifies which extensions under the Users tab are capable of belonging to a call qu eue by whether the Is Agent option is selected. The Is Agent option indicates tha t 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 lis t of agent s in the configuration for this option.

Figure 22: New Call Queue

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The Call Queues page, with the Queues tab select ed, 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, Ge ner a l Preferenc es to see which range of

numbers have been specified for ri ng gro ups.

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 suf ficiently descriptive as well . For example, “T ech nical Support” for the technical support que ue, “S al es”, and so o n .

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 ever y agent who isn’t on an active c all 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 a gent who has handled the fewes t 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 wit h Memory. It’s similar

to Round Robin, but smarter — it remembers over the course of days, weeks, or ye ars which agent rec eived the last c all 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 c an

receive calls as part of a call queue. All use rs 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 li sts all agent s and enable s 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 que ue 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 cal l queuing unt il af ter your cal l queue has been working for a while, and you have an idea of call volume and the turnover of calls by each agent.

 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 optio n allo ws ca ller s to ent er a que u e even if no

agents are logged into it. There are three options available:

– Yes - Callers can join a queue with no agents or only unavai lable

agents.

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– No - Callers can not join a queue with no agents. This is the default

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 grante d access to the queue. Ther e are thre e opt io n s avail ab le:

– 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,

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 forwar ded on to another agent.

 Wrapup Time - This is a buffer of time allowing your agents to f inish

work on one call and remain unavailable in the queue. The defa ult 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 complete.

 Max Len - This option sets the maximum number of callers allowed

in the queue befor e they are sent to vo ice mai l or rece ive a busy sign al. The default is “0,” which allows for an unlimite d number of calls in queue before they are se nt els ewh er e.

 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 fail s to answer a call, this option temporarily

postpones sending calls to that agent.

 Report Hold Time - The Report Hold T ime tells the agent how long

the call was holding i n que ue bef ore i t was s ent to the a gent. If the hol d time was short, the agent will probably be happy to accept the call. If the hold time wa s 10, 15, or 20 minutes, the agen t 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 Se tti ng s tab, accessible from the Manage Queues page, lets you specify the exten sions for agents to log into their queue, as well as callback logi n.

 Agent Login Extension - Use this f ield to specify the e xtension which

all agents can dial to log int o the q ueu e(s ) ass o ciat ed with their ex te nsion.

 Agent Cal lb ac k Log i n Ext e ns i on - Use this fie ld to specify the ext en-

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.

T o 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 underst and 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 Welc ome 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 ‘otherw is e’ & L isten fo r Key Pre s s  Play ‘to-reach-operator’ & Listen for KeyPress  Play ‘pls-h old-w h il e-t ry’ & L i sten fo r Key Pr es s  WaitExten ‘6’ Sec

In the example, whe n a c aller dials y our co mpany number ending in 7000, the call is answered, and after a pause of one second the caller is greeted in the foll owing manner: “ Thank you for calling. If you kn ow your par ty’s extension, please dial it now. Otherwise to reach an operator please dial

0.” If the caller tries an extens ion, 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 avail abl e menu options. The available menu options are as follows:

 Answer - This step is automatically added when creating a new menu.

This step answ er s the inco m ing cal l.

 Authenticate - The Authenticate step is used to restrict access to one

or more areas of your system. This is useful when one wants user s 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 use d to pla y an a udio fi le in the ba ckground

while waiting for the calle r to ente r 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 file s. In the example above, “Play ‘otherwise’ & Listen for KeyPress” is an example of using the Background option.

 Busytone - The Busytone option shoul d be selecte d if ther e is a step in

the process in which you want to play a busy signal to the call er. 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 calle r. You would play the congestion signal to the calle r, for instance, if you want to indicate the line is not avai lab le.

 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 tim e, the entry will be considered invalid. A field for entering the number of seconds before timeout appears next to the option.

 DISA - DISA (Direct In ward System Access) is an application which

allows caller s from outside the syst em to get acce ss to an intern a l dia l tone and place calls from within your internal system. A passcode is required. If the passcode ent ered 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 - I f a caller doe s not enter a respons e 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 sele ct. However, this option doe s not allow interr uption 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.

 WaitE xt en - The WaitExten option is specified to give a caller a

specified am oun t of time t o enter an ex ten sion .

 Goto Destination - The Goto Destination option lets a caller choose

to go to either a voice menu, a specific extens ion, 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 dif ferent 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 Span ish, all further voice menu prompts will be in Spanish (provided that lang uage module is loaded).

 Goto Directory - The Goto Directory option sends a ca ller 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 nu mber 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.

 Hangu p - The Hangup option terminates the call.  Custom App - This option allows you to specify a n Asterisk applica-

tion, along with the application’s corresponding parameters, which is not already listed in the Add new Step drop-do wn 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 title d Advanced Opti ons on page 138 for further details.

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Creating a Voice Menu

Use the following procedure as a guide to creat ing 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 Extensio ns 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 inb ound calle r to break o ut of the liste d Keypr ess Events an d dial another extensi on. A malic ious person may be able to hack through your Asterisk imple mentation to find an outside dial tone and use it for f raud. Any extension s that are known to t he 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 sy stem fro m being com pro m ised .

5. Specify the Keypress Event actions for digits 0-9 as well as *, #, t,

and i. The options available for a Keypress Eve nt are:

– None - The associated key is not enabled. – Goto Menu - Pressing a key with this option will send the cal ler to a

specified me nu.

– Goto Extensi on - Pressing a key with this option will send the

caller to a specified extension.

– Goto Qu eu e - Pressing a key with this option will send the caller to

the specified queue.

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– Operator - This option will se nd the caller to the designated opera-

tor.

– Hangup - Pressing a key with this option will terminate the call. – Congestion - Pressing a ke y with this option will play a busy signal. – Both the t key and i key should be used for specific actions. The

acti on as s ociated w it h t h e t key shoul d be th e desi red 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 constructe d your voice menu, cl ick 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 Pro mpts 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 use r 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 W AV for-

mat.

 Extensio n Us ed fo r Reco rding - 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 Prompt s

Click Upload a Voice Menu prompt to upload a custom voice menu prompt. You will be prompted to specif y t he path to t he audio file th at 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-rec ord the prompt, or delete the prompt. The prompts can now be included when creating voice menus.

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Time Intervals

Time inter vals 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 Tim e Interval definition is fairly simple . You just need to define a range of time in which you expect to receive cal ls. The following fields are used to cre at e the defi n itio n :

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 Time Interval Name - Specif y a u nique na me to he lp y ou identi fy thi s

time interval when it is referred to in the creation of calling rules. A name can be anything such as BusinessHours , Of f Hours, 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 “Business Hours” you would specif y Monday in the fir st 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 rad ial butt on if you wish t o speci fy a

day of a specific m onth inste ad of a day of a week. En ter th e day of the month, and then select the month from the drop-down list. For example, if you were creating a time interva l named Christmas, you would enter “25” and then select “December” from the drop-down list.

 Time - Y ou need to s pecify a tim e during whic h this i nterval should be

applied. Select either the Entire Day checkbox, or a St ar t Time and End Time. In the Business Hours example , whic h 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 definit ion 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 interva ls for each analog or VoIP trunk with which you receive inbound calls. Click Incoming Calling Rules to access the Incoming Calli ng Rules page.

Figure 28: Incoming Calling Rules

The main page displa ys the incoming calling r ules created for each tr unk. No rules are displayed if you have just setup your Asterisk Appli ance 50. Click New Incoming Rule to create a new incoming call ing 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.

 T run k - Select the t runk which t he incoming r ule sh ould appl y 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, holi day time, etc. You can also select None if you want to bypass any time intervals or patt erns.

 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, a nd 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 probably doesn’t require a long distance charge. Instead of adding a rule for every extension or phone number you call, spe cify the pattern in this rule similar to the example. All patterns begin with the unders core “_” 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 di git within the br ackets, in this instance 1, 2 , 3, 6,

7, 8, 9.

– . - The period is the wildcard which will match anyth ing 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 Extens ion by DID destination setting allows you to

route the incoming call to a local extension based on the DID (Direct Inward Dialing ) number tha t is sent to you by your telecommunications 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 rem ove a specif ied number of digits from the front of the DID number string be fore routing the call to a local extension.

Note: The Local Extension by DID destination setting is not applicable for analo g trunk s.

The rules you need to create are dependent on your needs. If you are configuring your syste m f or a b usine ss, f or example, you’ll pr obably 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. Cl i ck Apply Changes in the upper right corner of the page to make your changes immediate ly availab le. Each rule you create wil l be listed on the Incom i ng C all i ng R ul es page, or gani zed 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 spec ify whether voicemail is enabl ed for an extension, as well as the PIN for retrieving voic email. 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 pa ge is the primary page used to configure Asterisk Appliance 50 voicemail. Standard configuration inf ormation is present, allowing you to confir m the extension used to chec k messages, as well as general parameters such as the following:

 Extensio n fo r Checking Messa g es - 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 voice-

mail 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 specif y 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 salutation, or after recording the message. If this option is not enabled, a caller’s pr essing “0” will be ignored.

There are several options which can define the characteristics of the voicemail messages in the syste m.

 Maximum Messages per Folder - This fiel d sets the maximum num-

ber of messages that a user can have in any over their voice mail box folders.

 Maximum Message Time - The maxim um 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 le ft 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 Messa ge Caller ID option r eads the

caller ID before the voice mail message is played.

 Say Message Duration - If this option is enabled the duration of the

message, in m inut es , wi ll be play ed b ack befo re t he voi cem ai l 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 rerecord it.

E-mail Set ti ngs

The E-mail Settings page is used to set e-mail options for voicemail, as well as the format of the e-m ai l s 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

voicemai l is sent to a users e-mail address as an att achment. Click the check box to enable this option. Messages will be sent in the .wav format.

 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. T o 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 Set tin gs

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 Applianc e 50 can connect to, without authentication, 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 all ows 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/Inter co m Group to de fine which available users wil l 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 initia ting 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 inte rcom call is connected to inf or m use rs that they can begin talking.

 Page/Intercom Group Members - This is the list of available users

which are part of this page/intercom group.

 Availabl e us er s - This is the list of users which are available to be

assigned to this page/in tercom group.

The double lef t arrows will move a ll 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/inte rcom 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/interc om 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 Ex te ns i on as in tercom - Sp ecify 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 f or paging

and intercom.

Figure 34: Paging & Intercom Settings

The following setting is available:

 Alert-Info He ad er - T h is is the v alue tha t is sen t i n t he aler t 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 chang es, 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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Digium AA50 Administrator's Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Introduction to Asterisk Appliance 50 Documentation

Chapter 1 Overview

Features:

Chapter 2 Unit Installation

Unpacking the Unit

Inspecting Your Shipment

Identifyi ng C om m un ic a tion Por ts

Underst andi n g th e LEDs

Using the Configuration Reset Switch

Installin g the A st eris k Appliance 50

Mounting the Asterisk Appliance 50

Chapter 3 Telephone System Configuration

Log On to the Asterisk Appliance 50

The Asteris k Appliance 50 Inte rface

Analog Hardware Configuration

Trunk C onfiguration

Analog Trunks

Outgoing Calling Rules

Dial Plans

User Extens i ons

Ring Groups

Music on Hold

Call Queu es

Agent Login Settings

Voice Menus

Record a Voice Menu

Time Intervals

Incoming Calling Rules

Voicemail

Paging/Intercom