Digium Asterisk Appliance 50 AA50 Administration Manual

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Asterisk Appliance™ 50 (AA50)
Administrator's Manual
Release 1.05

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Summary of Contents for Digium Asterisk Appliance 50 AA50

  • Page 1 Asterisk Appliance™ 50 (AA50) Administrator’s Manual Release 1.05...
  • Page 2 Digium, Inc. Digium, Inc. has made every effort to ensure that the instructions contained in this document are adequate and error free. The manufacturer will, if necessary, explain issues which may not be covered by this documentation.
  • Page 3 (Annex C) Telecom: FCC Part 68, TIA-968 AS/ACIF S002 TBR-21 1998 Industry Canada IC-CS-03 EMC: FCC Part 15 Class A EN55022/CISPR22 Class A EN55025 IEC 61000 AS/NZS CISPR22 Class A CNS13438 VCCI V-32005.04 Release 1.05 Digium, Inc. Page 3...
  • Page 4 If you experience problems with the Asterisk Appliance, contact Digium, Inc. (+1.256.428.6161) for repair and/or warranty information. If the equipment is causing harm to the telephone network, the telephone company may request that you disconnect the equipment until the problem is resolved.
  • Page 5 This device complies with part 15 of FCC rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) This device must accept any interference received, including interference that may cause undesired operation. Release 1.05 Digium, Inc. Page 5...
  • Page 6 Explains resolutions to common problems and frequently asked questions pertaining to the unit installation and usage. Pin Assignments Lists the connectors and pin assignments. Specifications Details unit specifications. Glossary and Defines terms related to this product. Acronyms Release 1.05 Digium, Inc. Page 6...
  • Page 7 (ESD) wrist strap while handling the device. The Electrical Hazard Symbol indicates a possibility of electrical shock when operating this unit in certain situations. To reduce the risk of damage or injury, follow all steps or procedures as instructed. Release 1.05 Digium, Inc. Page 7...
  • Page 8: Important Safety Instructions

    Refer servicing to qualified service personnel. Caution. This unit must be used in restricted access locations and installed in UL Listed I.T.E. only. Release 1.05 Digium, Inc. Page 8...
  • Page 9 Service providers to which 911 or Emergency calls are placed should ensure their provider properly forwards the customer's accessible PSTN phone number to the emergency call handling center. Save these instructions for future reference. Release 1.05 Digium, Inc. Page 9...
  • Page 10: Table Of Contents

    Adding Service Providers ....... . 45 Release 1.05 Digium, Inc. Page 10...
  • Page 11 Glossary and Acronyms ........79 Release 1.05 Digium, Inc. Page 11...
  • Page 12 Update ........70 Figure A-1: Back Panel Ports ......73 Release 1.05 Digium, Inc. Page 12...
  • Page 13 LAN & WAN Ethernet Port Pinouts ....76 Table B-4: Maximum 12V Power Consumption ... . . 78 Release 1.05 Digium, Inc. Page 13...
  • Page 14: Overview

    Chapter 1 Overview ® The Digium Asterisk Appliance 50 (AA50) is a stand alone PBX which runs a version of Asterisk. It is suitable for the desktop, or mounting in a typical network closet or restricted access location. The AA50 is ideal for small office environments or as an extension to a central Asterisk PBX.
  • Page 15: Features

    High performance Analog Devices Incorporated (ADI) BlackFin BF537 processor uClinux Operating System Transcoding provided on the Blackfin processor 32ms of analog port echo cancellation 8MB on board serial Flash memory 64MB 16 bit parallel SDRAM Front panel LEDs Release 1.05 Digium, Inc. Page 15...
  • Page 16: What Is Asterisk

    PBX. You can also utilize the appliance as an adjunct application server for conferencing, voice mail, or IVR. Asterisk will work with most standards-based IP telephone handsets and software. Analog phones and ADSI-screen phones are also supported. Release 1.05 Digium, Inc. Page 16...
  • Page 17: Unit Installation

    Identifying Communication Ports on page 19 Understanding the LEDs on page 19 Using the Configuration Reset Switch on page 23 Installing the Appliance on page 24 Mounting the Asterisk Appliance on page 26 Figure 1: The Asterisk Appliance Release 1.05 Digium, Inc. Page 17...
  • Page 18 If damage is suspected, file a claim with the carrier and contact your reseller from which the unit was purchased or Digium Technical Support (+1.256.428.6161). Keep the original shipping container to use for future shipment or proof of damage during shipment.
  • Page 19 There are 15 LEDs on the front panel of the Asterisk Appliance. 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. Release 1.05 Digium, Inc. Page 19...
  • Page 20 Link is up at 100Mbps. LED flashes at 1/ (flashing) 10 second intervals as traffic is detected. Orange Link is up at 10Mbps. LED flashes at 1/10 (flashing) second intervals as traffic is detected. Release 1.05 Digium, Inc. Page 20...
  • Page 21 Red (solid) Port is configured for FXO operation and is enabled. A telephone line may be connected to this port. Telephone line is ringing. (flashing) Red (slow Telephone line is in use. blinking) Release 1.05 Digium, Inc. Page 21...
  • Page 22: Figure 2: Example Asterisk Appliance Port Identification

    Chapter 2: Unit Installation Figure 2: Example Asterisk Appliance Port Identification Release 1.05 Digium, Inc. Page 22...
  • Page 23: Figure 3: Asterisk Appliance Back View

    Caution. Pressing the RST CFG switch will cause loss of all configuration settings and reset administration passwords. Power Supply Ports Craft Port Configuration Analog Reset Switch Ports Port Figure 3: Asterisk Appliance Back View Release 1.05 Digium, Inc. Page 23...
  • Page 24 Appliance will immediately power on once connected to a power source. When the unit completes the boot process, the left-most eight LEDs will indicate how the analog ports are configured. Red indicates FXO, Green for FXS, and off if not installed. Release 1.05 Digium, Inc. Page 24...
  • Page 25 You are now ready to configure your Asterisk Appliance via the GUI. Caution. This unit must be connected to the Telecommunications Network in your country using an approved line cord, e.g.: for Australia use only line cords complying with ACA Technical Standard TS008. Release 1.05 Digium, Inc. Page 25...
  • Page 26: Figure 4: Mounting Instructions

    Figure 4 below illustrates the proper mounting installation options: Figure 4: Mounting Instructions Warning. Do not place anything (including paper) on top of the Asterisk Appliance. To allow proper cooling, these units must not be stacked. Release 1.05 Digium, Inc. Page 26...
  • Page 27 Leave approximately 1/4-inch of the screw protruding from the wall to allow the head of the screws to slide into the keyed insets, mounting the unit to the wall. Release 1.05 Digium, Inc. Page 27...
  • Page 28: Telephone System Configuration

    After connecting to the Asterisk Appliance, 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. Release 1.05 Digium, Inc. Page 28...
  • Page 29: Figure 5: Gui Login

    The first time you log on you will be prompted to change your password from the default. You will then be prompted to log on with the new password. Once the log on process is complete the AsteriskGUI home page will be displayed. Release 1.05 Digium, Inc. Page 29...
  • Page 30: Figure 6: Home Page

    For example, the users configured in the Users tab are used to assign voice mail options on the Voicemail tab and can be included in the Call Queues. Release 1.05 Digium, Inc. Page 30...
  • Page 31 As noted above, these changes will not become permanent until you have clicked Save Configuration on the home page. Click Logout on any page to exit the Asterisk Appliance interface. Release 1.05 Digium, Inc. Page 31...
  • Page 32: User Extensions

    Chapter 3: Telephone System Configuration User Extensions Click the User Extensions tab and you'll see the extensions you created in the initial setup process. Figure 7: User Extensions Release 1.05 Digium, Inc. Page 32...
  • Page 33 Phone Serial - The Phone Serial field is used to specify the MAC address of the SIP phone connected to the port specified above. The MAC address associates the phone with this extension and enables the synchronization of provisioning information. Release 1.05 Digium, Inc. Page 33...
  • Page 34 An extension listed as Is Agent can be added to queues from the CallQueues option on the left toolbar. To add new extensions, select New Entry in the area under Extensions and then click New. The system defaults to four-digit extensions, beginning with the number 6000. Release 1.05 Digium, Inc. Page 34...
  • Page 35: Conferencing

    Every company reaches the point of needing more people on a phone call than it can effectively include through three-way calling. Asterisk conference bridges allow you to include more people as well as project a professional image. Figure 8: Conferencing Release 1.05 Digium, Inc. Page 35...
  • Page 36 If you are bringing together sales teams or vendors on a conference, it is preferable to keep them from chatting amongst themselves before the host arrives. These features allow you to handle the following requirements: Release 1.05 Digium, Inc. Page 36...
  • Page 37 The main conference extension of 6003 is configured with Wait for Marked User selected. Everyone in the conference arriving from extension 6003 remains silent until the CEO arrives. Release 1.05 Digium, Inc. Page 37...
  • Page 38: Voicemail

    The Voicemail Configuration page displays all the extensions to the left, including voice mail. Standard configuration information is also present, allowing you to confirm the extension used to check messages, as well as general parameters such as the following: Release 1.05 Digium, Inc. Page 38...
  • Page 39 Minimum Message Time - The minimum duration of a message is dictated here. Any message left that’s under the listed duration is dis- carded and isn’t processed or retrievable. Release 1.05 Digium, Inc. Page 39...
  • Page 40 Standard options are presented to you, allowing you to discard the message or re-record it if you aren’t happy with it. Release 1.05 Digium, Inc. Page 40...
  • Page 41: Calling Queues

    Is Agent, that extension won’t appear in the list of agents in the configuration for this option. The following illustration shows the Queue Extension Configuration window displaying the available options. Figure 10: Call Queues Release 1.05 Digium, Inc. Page 41...
  • Page 42 Fewest Calls - The agent who has handled the fewest calls receives the next incoming call. Random - Goes by the luck of the draw; any agent can receive the – next incoming call. Release 1.05 Digium, Inc. Page 42...
  • Page 43 The default is “0,” which allows for an unlimited number of calls in queue before they are sent elsewhere. Release 1.05 Digium, Inc. Page 43...
  • Page 44 If the hold time was 10, 15, or 20 minutes, the agent might want to brace for a frustrated customer, but at least the agent isn’t overwhelmed. Release 1.05 Digium, Inc. Page 44...
  • Page 45: Adding Service Providers

    VoIP provider as service providers. The service providers may use analog lines for your local carrier or IP connections for your VoIP carriers. Figure 11: Service Providers Release 1.05 Digium, Inc. Page 45...
  • Page 46: Figure 12: Add New Service Provider

    Service Providers window. To add a new provider, click Add Service Provider. The Add Service Provider dialog box will be displayed similar to the following illustration. Figure 12: Add New Service Provider Release 1.05 Digium, Inc. Page 46...
  • Page 47 – Host - The IP address of your service provider. Username - The user name associated with your provider account. – Password - The password associated with your provider account. – Release 1.05 Digium, Inc. Page 47...
  • Page 48 Com- ment. Insecure - This option specifies how connections to a service pro- – vider (host) should be handled. Valid options are very|yes|no|invite|port. The default is no (authenticate all connec- tions). Release 1.05 Digium, Inc. Page 48...
  • Page 49 You can then test your changes to insure that they work. If you are happy with the results, return to the Home page and click Save Configuration to make the changes permanent. Release 1.05 Digium, Inc. Page 49...
  • Page 50: Calling Rules

    Figure 13: Calling Rules The Calling Rules menu shows every rule name established, a brief description of the rule name, as well as the service provider used to complete the specific type of call. Release 1.05 Digium, Inc. Page 50...
  • Page 51: Figure 14: New Calling Rule

    9 followed by 7 digits, it would be a local call that proba- bly didn’t require a long distance charge. Instead of adding a rule for every extension or phone number you call, specify the pattern in this rule similar to the example. Release 1.05 Digium, Inc. Page 51...
  • Page 52 Click New at the top of the Calling Rules page and create a new dial plan name. You can then add calling rules for that dial plan definition. Release 1.05 Digium, Inc. Page 52...
  • Page 53: Incoming Calls

    Figure 15: Incoming Calls Some example incoming rules are show in the figure above. Click Add a Incoming Rule to create a new incoming rule. There are only a few options you will need to define. Release 1.05 Digium, Inc. Page 53...
  • Page 54 Once you have completed the definition of your incoming call rules, click Save. Click Activate Changes to test the functionality of your new rules. Click Save Configuration on the Home page if the rules work according to your needs. Release 1.05 Digium, Inc. Page 54...
  • Page 55: Networking

    Chapter 3: Telephone System Configuration Networking The Networking page is used to configure your general network settings, as well as your Wide Area Network (WAN) and local network (LAN) settings. Figure 16: Networking Release 1.05 Digium, Inc. Page 55...
  • Page 56 NTP server. SSH - Select the SSH checkbox to activate the SSH server on the Appliance. The default root password is digium. Enabling this option will cause your unit to provide SSH access on both WAN and LAN interfaces, which can pose a security risk.
  • Page 57 You can then test your changes to insure that they work. If you are happy with the results, return to the Home page and click Save Configuration to make the changes permanent. Release 1.05 Digium, Inc. Page 57...
  • Page 58: Voice Menus

    The best way to understand how a voice menu is constructed is to examine the default “Welcome” menu provided with the Asterisk Release 1.05 Digium, Inc. Page 58...
  • Page 59 6 seconds. This is an example of a basic voice menu. In the example, each action is a step chosen from the list of available menu options. The available menu options are as follows: Release 1.05 Digium, Inc. Page 59...
  • Page 60 If a full extension is not entered in the specified time, the entry will be considered invalid. A field for entering the number of seconds before timeout appears next to the option. Release 1.05 Digium, Inc. Page 60...
  • Page 61 Dial RingGroup - This option will dial a specified RingGroup. For example, if your menu says “Press 1 for Technical Support”, the Tech- nical Support ring group will be dialed. Hangup - The Hangup option terminates the call. Release 1.05 Digium, Inc. Page 61...
  • Page 62 Any extensions that are known to the public should be completely handled by the Keypress Events; callers should not be allowed to dial other extensions. Sticking to this policy protects your Asterisk system from being compromised. Release 1.05 Digium, Inc. Page 62...
  • Page 63 Once you have constructed your voice menu, click Save. You can then click Activate Changes to add the voice menu to your current configuration. If you are happy with the voice menu, click Save Configuration on the Home page. Release 1.05 Digium, Inc. Page 63...
  • Page 64: Record A Menu

    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. Release 1.05 Digium, Inc. Page 64...
  • Page 65: Call Parking

    What Extensions to Park Calls On - The extensions specified here will be the “parking lot” designations for the calls you place on hold. The call on hold will be retrieved by dialing one of these extensions. Release 1.05 Digium, Inc. Page 65...
  • Page 66 Note: In order to properly park a call, you must use attended transfer functions. Using a blind transfer function will not provide the parking number to the person parking the call. This makes recovery of the call impossible, except for the fall through timeout. Release 1.05 Digium, Inc. Page 66...
  • Page 67: Ring Groups

    To create a new ring group click New Ring Group at the top of the Ring Groups page. Figure 20: Ring Groups Note: You will not be able to define a ring group without user extensions or trunks defined. Release 1.05 Digium, Inc. Page 67...
  • Page 68 Lastly, determine what action you want the system to take if no one answers the call. You can either direct the call to the voicemail of the first user, go to an IVR menu, or end the call. Release 1.05 Digium, Inc. Page 68...
  • Page 69: Asterisk Management Options

    Setup Wizard. The Advanced/Basic tab allows you to enable or disable advanced options. The basic options are displayed by default. Please refer to the Advanced Options section for a description of the advanced options. Release 1.05 Digium, Inc. Page 69...
  • Page 70: Update

    The Updates tab provides an interface for downloading or uploading newer firmware images to the Appliance. Customers with an active Service Subscription may visit the Digium.com website to register their appliance, activate their Service Subscription, and download software updates for the appliance. The Updates tab provides two interfaces for putting a new image on the appliance.
  • Page 71: Advanced Options

    Global SIP Settings - The SIP configuration settings can be enabled or disabled from this page. Global IAX Settings - The IAX configuration settings can be enabled or disabled from this page. Release 1.05 Digium, Inc. Page 71...
  • Page 72: Chapter 4 Troubleshooting

    #asterisk on (irc.freenode.net). Maintenance Support Program Digium is dedicated to supporting your Asterisk system by offering full technical support through our Maintenance Support Program. The Maintenance Support Program covers all Digium hardware and the complete Asterisk software suite. Through this program, you can be at ease knowing that your business will always have access to the Asterisk experts.
  • Page 73: Appendix A Pin Assignments

    This appendix provides pin assignments for the Asterisk Appliance unit. Ports Craft Port Port Analog Ports Figure A-1: Back Panel Ports All eight analog ports on the Asterisk Appliance are 8-pin RJ11 ports. The pin assignments are identified in Table A-2. Release 1.05 Digium, Inc. Page 73...
  • Page 74: Table A-1: Craft Port Pinout

    (Connect to DB9 Pin 7) Open RTS (From Appliance) (Connect to DB9 Pin 8) Note: The secondary serial port is not wired in the craft port adapter that may be supplied with the unit. Release 1.05 Digium, Inc. Page 74...
  • Page 75: Table A-2: Rj11 Analog Port Connector

    Appendix A: Pin Assignments Table A-2: RJ11 Analog Port Connector Diagram Description Unused Pin 1 Pin 6 Unused Ring Unused Unused Release 1.05 Digium, Inc. Page 75...
  • Page 76: Table A-3: Lan & Wan Ethernet Port Pinouts

    Appendix A: Pin Assignments Table A-3: LAN & WAN Ethernet Port Pinouts Diagram Description Rx Receive Negative Pin 1 Pin 8 Rx Receive Positive Tx Transmit Negative Unused Unused Tx Transmit Positive Unused Unused Release 1.05 Digium, Inc. Page 76...
  • Page 77: Specifications

    DC Power - 6.3mm O.D., 2mm pin; 12V 3A center positive CompactFlash - Type 1 Environment. Temperature: 0 to 40° C (32 to 104° F) operation -20 to 70° C (4 to 158° F) storage Humidity: Up to 90% non-condensing Release 1.05 Digium, Inc. Page 77...
  • Page 78: Table B-4: Maximum 12V Power Consumption

    36 Watts Each FXS port in use with 3REN load 1.5 Watts Note: Power consumption is determined by the number of analog phones connected to the FXS ports and the REN rating of the phones. Release 1.05 Digium, Inc. Page 78...
  • Page 79 The smallest element of information in a digital system. A bit can be either a zero or a one. bits per second A measurement of transmission speed across a data connection. Release 1.05 Digium, Inc. Page 79...
  • Page 80 CLEC competitive local exchange carrier A term for telephone companies established after the Telecommunications Act of 1996 deregulated the LECs. CLECs compete with ILECs to offer local service. See also LEC and ILEC. Release 1.05 Digium, Inc. Page 80...
  • Page 81 Digital Signal, Level 3 T3 in North America and Japan, E3 in Europe. Up to 672 voice channels (DS0s). DS3/T3/E3 lines are not part of the PSTN DTMF Dual Tone Multi-Frequency Push-button or touch tone dialing. Release 1.05 Digium, Inc. Page 81...
  • Page 82 The International Telecommunications Union recommendation for an algorithm designed to transmit and receive mulaw PCM voice and A-law at digital bit rate 64 Kbps. This algorithm is used for digital telephone sets on digital PBX. Release 1.05 Digium, Inc. Page 82...
  • Page 83 The LECs that were the original carriers in the market prior to the entry of competition and therefore have the dominant position in the market. interface A point of contact between two systems, networks, or devices. Release 1.05 Digium, Inc. Page 83...
  • Page 84 FDM separates signals by dividing the data onto different carrier frequencies, and TDM separates signals by interleaving bits one after the other. multiplexer A device which transmits multiple signals over a single communications line or channel. See multiplexing. Release 1.05 Digium, Inc. Page 84...
  • Page 85 A communications network which uses telephones to establish connections between two points. Also referred to as the dial network. quality of service A measure of telephone service, as specified by the Public Service Commission. Release 1.05 Digium, Inc. Page 85...
  • Page 86 TDM separates signals by interleaving bits one after the other. telco A generic name which refers to the telephone companies throughout the world, including RBOCs, LECs, and PTTs. Release 1.05 Digium, Inc. Page 86...
  • Page 87 Voice over IP Technology used for transmitting voice traffic over a data network using the Internet Protocol. Zaptel (Zap) Zapata Telephony Project dedicated to implementing a reasonable and affordable Computer Telephony platform into the world marketplace. Release 1.05 Digium, Inc. Page 87...

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