PIKA WARP the Appliance for Asterisk / FreePBX · PDF file4 Overview FreePBX® is an easy to use Web GUI (graphical user interface) that controls and manages Asterisk. The FreePBX

PIKA WARP the Appliance for

Asterisk / FreePBX

User Guide

Copyright (c) 2010. All rights reserved.

Table of Contents

1 Copyright Information 1

2 Contacting PIKA Technologies 2

3 Introduction 3

3.1 Related Documents 4

4 Overview 5

5 Using FreePBX 6

6 Logging on to FreePBX 8

6.1 Changing the Default Passwords 9

7 Adding Trunks 12

8 Adding Extensions 17

9 Conferencing 22

10 Setting Up an Interactive Voice Response System 24

11 Adding Routes 26

12 System Settings 33

13 Fax 39

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14 Module Admin 44

15 Backup and Restore 46

16 Miscellaneous 48

16.1 Using the Asterisk CLI 48

16.2 SD Card Usage 48

16.3 System Recordings 50

16.4 Customizing the FreePBX Logo 53

16.5 General Settings 53

17 Channel Drivers 55

17.1 GSM Channel Driver 55

17.1.1 GSM Configuration 55

17.1.2 Using the GSM CLI 58

17.2 PIKA Channel Driver 61

17.2.1 Configuration 61

17.2.2 Using the PIKA CLI 75

17.3 Channel Status on the LCD 76

17.3.1 BRI Link State Display 79

18 Troubleshooting 81

19 FreePBX FAQ 82

Index a

PIKA WARP the Appliance for Asterisk /

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1 Copyright Information

COPYRIGHTS

Copyright 2010 PIKA Technologies Inc.

TRADEMARKS

PIKA is a registered trademark of PIKA Technologies Inc. All other trademarks, product names and company names

and/or logos cited herein, if any, are the property of their respective holders.

DISCLAIMER

This document is provided to you for informational purposes only and is believed to be accurate as of the date of its

publication, and is subject to change without notice. PIKA Technologies Inc. assumes no responsibility for any errors or

omissions in this document and shall have no obligation to you as a result of having made this document available to

you or based upon the information it contains.

1 PIKA WARP the Appliance for Asterisk /

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2 Contacting PIKA Technologies

Customer Care

For support issues, phone or e-mail our Customer Care department at the following:

Tel: +1-613-591-1555

FAX: +1-613-591-9295

Email: [email protected]

International Headquarters

PIKA Technologies Inc.

535 Legget Drive, Suite 400

Ottawa, Ontario, Canada K2K 3B8

Tel: +1-613-591-1555

FAX: +1-613-591-9295

Email: [email protected]

Internet

Visit our website at www.pikatechnologies.com for the latest news, product announcements, downloads, online

community, documentation updates, and contact information.

PIKA WARP the Appliance for Asterisk / 2

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http://www.pikatechnologies.com

3 Introduction

PIKA WARP the Appliance for Asterisk® / FreePBX® provides an integrated PBX solution for system integrators and

Asterisk resellers who are familiar with FreePBX, Elastix, Trixbox CE and similar products. PIKA has eliminated all the

integration issues that you may have experienced in the past with PCs (obsolete hard drives, motherboards etc).

Whether your customers require SIP only configurations or prefer the safety of added PSTN connectivity, WARP for

Asterisk / FreePBX with your optionally added application for differentiation is ideal for the SMB market.

To speed your time to market, PIKA provides the appliance with Asterisk and FreePBX pre-installed. All integration

testing is done for you while still providing the flexibility to add your own differentiation through custom branding.

Asterisk is a registered trademark of Digium Inc. (Copyright © 2010 All rights reserved).

FreePBX is a registered trademark of Bandwidth.com (Copyright © 2010 All Rights Reserved)

Guide Organization - This guide is organized as follows:

• Introduction

• Overview ( pg. 5) - Describes the features of Warp for Asterisk / FreePBX

• Using FreePBX ( pg. 6) - Describes the layout and use of FreePBX page

• Logging on to FreePBX ( pg. 8) - Describes how to log on to FreePBX and change default passwords

• Adding Trunks ( pg. 12) - Describes how to add PIKA FXO and BRI trunks

• Adding Extensions ( pg. 17) - Describes how to add extensions for PIKA FXS lines

• Conferencing ( pg. 22) - Describes how to set up conferences

• Setting Up an IVR ( pg. 24) - Describes how to set up an IVR

• Adding Routes ( pg. 26) - Describes how to add outbound and inbound routes using PIKA channels

• System Settings ( pg. 33) - Describes the appliance system administration pages

• Fax ( pg. 39) - Describes how to send a fax using the PIKA fax module

• Backup and Restore ( pg. 46) - Describes important information about backing up files when upgrading the

appliance software

• Miscellaneous ( pg. 48) - Describes other useful information about FreePBX pages and PIKA customizations

• Channel Drivers ( pg. 55) - Describes configuration settings and CLI commands for the GSM and PIKA channel

drivers

• FreePBX FAQ ( pg. 82) - Answers to common questions about using FreePBX

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3.1 Related Documents

The following documents are related to the PIKA WARP for Asterisk/FreePBX Guide. These documents are linked

together and constitute the complete set of documentation for PIKA WARP the Appliance. All documents are available

at http://www.pikatechnologies.com/appliancedownloads.

PIKA WARP the Appliance User Guide: Describes installation and configuration of the appliance.

PIKA Application Development Suite (PADS) User Manual: This guide describes the software development kit to

develop software for the appliance.

PIKA WARP the Appliance Hardware Manual: This manual describes the base board and plug-in modules.

PIKA WARP the Appliance Release Notes: These notes describe the contents of the release, including known

product issues.

PIKA WARP the Appliance for Asterisk / 3.1 Related Documents

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http://www.pikatechnologies.com/appliancedownloads

4 Overview

FreePBX® is an easy to use Web GUI (graphical user interface) that controls and manages Asterisk. The FreePBX

implementation on the WARP appliance uses the FreePBX source code directly from the FreePBX Web site at

www.freepbx.org. All updates and additions to the GUI come from the FreePBX module repository. PIKA has made

minor modifications to support FreePBX on the embedded platform, however, the goal is to limit modifications by

PIKA to the FreePBX source code.

PIKA adds modules to FreePBX using the plug-in module architecture. The modules are free and you can use and

modify them as needed. To support FreePBX on the appliance, PIKA adds the required applications such as Asterisk,

MySQL, PHP, Perl, lighttpd, and auto-install scripts.

Warp for Asterisk / FreePBX has the following features:

• Core FreePBX module set

• Common modules (from FreePBX module repository):

• IVR module

• Find Me module

• Conferencing module

• Manager (Asterisk API module)

• Backup/Restore module

• PIKA Modules (via plug-in module architecture):

• System Tools for the WARP appliance

• hardware and software information

• network configuration

• email server configuration

• appliance restart

• PIKA channel status

• Fax for the WARP appliance

For more information, visit:

• PIKA Forum - support and information

• WARP Xtra - WARP developers community Web site

• http://www.freepbx.org/support/documentation - Documentation on the FreePBX Web site


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http://forum.pikatechnologies.com

http://pikawarp.org

http://www.freepbx.org/support/documentation

5 Using FreePBX

The FreePBX Web pages follow a standard layout. The figure below shows a typical FreePBX Web page.

To activate any configuration changes, you must click Apply Configuration Changes below the navigation tabs at the

top of the page. This link is not visible unless you make changes. The following dialog box appears.


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Click Continue with reload to confirm the changes.

The following message appears. When the message disappears, Asterisk uses the new settings for new calls, active calls

are not affected.

FreePBX has been tested with the following browsers:

• Google Chrome (best performance)

• Firefox

• Internet Explorer

• Safari (performance may be slow)


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6 Logging on to FreePBX

To log on to the FreePBX GUI, you need the IP address of the appliance. To obtain the IP address, after the appliance

starts, press the touch sensitive button to the right of the LCD screen. The button toggles the LCD display between the

line status view and the IP address view. Type the IP address of the appliance in the address bar of any Web browser.

The FreePBX Welcome page appears.

Click the FreePBX Administration link. The following dialog box appears.

Type the user name and password in the appropriate text box. The default user name is admin and the default

password is admin. Click the OK button. The FreePBX Status page appears.


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• The System Statistics section shows hardware status information: memory usage, CPU usage, disk space usage, and

network status.

• The Server Status section shows the status of the main software components on the appliance.

• The Uptime section shows the in-service time for the appliance and Asterisk. The Last Reload field shows when the

Asterisk configuration last changed.

• The FreePBX statistics section shows call status information, including IP phone registration.

• The FreePBX Notices section shows information about the FreePBX installation and configuration. Click the icon

beside an item for more information about the item.

6.1 Changing the Default Passwords

For security reasons, we recommend that you change the passwords for all users with administrator level permissions.

Changing the Password for the Appliance root User

The appliance has a default root user account and the corresponding password is pikapika.

Use an SSH client to open a session to the appliance using the IP address displayed on the LCD. Type the default user

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name and password at the prompts. If you want to change the password for the root user account, type passwd root at

the Linux command prompt on the appliance. Follow the prompts to enter and confirm the new password. Password

information is preserved between restarts.

Changing the Password for the FreePBX admin User

To change the password for the admin user, in the left navigation menu, click Setup and click Administrator. The

Add Admin Web page appears. Click the admin button on the right side of the page. The following Web page appears.

Type the new password in the Password text box. Optionally, change any other settings as needed. Click Submit

Changes when you finish. To apply your changes, click Apply Configuration Changes at the top of the page.

Changing the Password for the Database Administrator

To change the FreePBX database administrator password, perform the following steps.

PIKA WARP the Appliance for Asterisk / 6.1 Changing the Default Passwords

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1. Update the database:

• At the Linux command prompt on the appliance, type mysql --user=root --password=amp109

• At the mysql prompt, type the following commands, where newpassword is the password you want to use:

• UPDATE mysql.user SET Password=PASSWORD('newpassword') WHERE User='asterisk';

• FLUSH PRIVILEGES;

• exit

2. Update amportal.conf:

• In the file /persistent/etc/amportal.conf, change the line shown below. Type the password you want to use in

place of the default password, amp109. It must match the new password you used in the previous step.

AMPDBUSER=asterisk

AMPDBENGINE=mysql

AMPDBPASS=amp109

3. Apply the change to amportal.conf:

• At the Linux command prompt, type /usr/src/freepbx/apply_conf.sh

4. In the file /persistent1/var/www/html/admin/functions.inc.php, change the line shown below. Type the password

you want to use in place of the default password, amp109. It must match the password in the file

/persistent/etc/amportal.conf.

'AMPDBHOST' => array('std' , 'localhost'),

'AMPDBUSER' => array('std' , 'asteriskuser'),

AMPDBPASS' => array('std' , 'amp109'),

5. Restart Asterisk:

• At the Linux command prompt, type amportal restart.

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7 Adding Trunks

This section describes how to set up trunks in Asterisk for PIKA BRI, FXO, and GSM channels.

To add a trunk in Asterisk, click the Setup tab at the top of the left navigation menu. In the left navigation menu, click

Trunks. The following Web page appears.

Adding PIKA FXO and BRI Trunks

PIKA FXO and BRI trunks are both custom trunks in Asterisk.

Click the link Add Custom Trunk on the Add a Trunk Web page. The following Web page appears.


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In the text box Trunk Description, type a descriptive name for the trunk. In the Custom Dial String text box, type

the dial string for a PIKA trunk. Optionally, assign values to any other fields as needed. Position the mouse pointer over

the field name for information about the field. Click Submit Changes when you finish. To apply your changes, click

Apply Configuration Changes at the top of the page.

Dial Strings for FXO Trunks

We recommend that you use FXO channels as a group. In the Custom Dial String text box, type

PIKA/fxo/gnumber/$OUTNUM$ where number is the group number of the FXO module(s). To confirm the number,

visit the Warp Status Page (click the Tools tab at the top of the left navigation menu and click the Show Channels link

in the left navigation menu). The sample output below shows an FXO group where the group number is zero. The

corresponding dial string is PIKA/fxo/g0/$OUTNUM$.

$OUTNUM$ is used when placing a call to an external number. If you use any prefix digits, $OUTNUM$ is the

number after the prefix digits are removed. Prefix digits are normally part of a dial pattern for an outbound route. Refer

to section Adding Routes ( pg. 26) for more information.


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You can use the following letters to specify the search algorithm to use when selecting a free channel in the group:

• g - choose the first free channel, searching from lowest to highest channel number

• G - choose the first free channel, searching from highest to lowest channel number

• r - use round-robin channel selection, starting with the lowest channel number

• R - use round-robin channel selection, starting with the highest channel number

WARP Channel Status

Group: 0, type FXO: channels: 1 to 4

PIKA Channel Asterisk Group: 0, type FXO: channels: 1 to 4

Channel number: 1, asterisk group: 0 type: FXO, state = CHANNEL IS DOWN

Channel number: 2, asterisk group: 0 type: FXO, state = CHANNEL IS READY

Channel number: 3, asterisk group: 0 type: FXO, state = CHANNEL IS DOWN

Channel number: 4, asterisk group: 0 type: FXO, state = CHANNEL IS READY

Group: 1, type FXS: channels: 1 to 5

PIKA Channel Asterisk Group: 1, type FXS: channels: 1 to 5

Channel number: 1, asterisk group: 1 type: FXS, state = CHANNEL IS READY





Group: 3, type AUDIO: channels: 1 to 1

PIKA Channel Asterisk Group: 3, type AUDIO: channels: 1 to 1

Channel number: 1, asterisk group: 0 type: AUDIO, state = CHANNEL IS READY

If you want to use a specific channel, in the Custom Dial String text box, type PIKA/fxo/number/$OUTNUM$

where number is a value between one and eight, depending on how whether you have one or two FXO modules installed

on the appliance. Each module has four ports.

Dial Strings for BRI Trunks

BRI channels should be added as PIKA/digital/gnumber/$OUTNUM$ where number is the group number of the BRI

span. To confirm the number, visit the Warp Status page (click the Tools tab at the top of the left navigation menu

and click the link Show Channels in the left navigation menu). The following sample output shows two BRI groups.

The corresponding dial string is PIKA/digital/g1/$OUTNUM$.

You can use the following letters to specify the search algorithm to use when selecting a free channel in the group:



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WARP Channel Status

Group: 0, type FXS: channels: 1 to 5

PIKA Channel Asterisk Group: 0, type FXS: channels: 1 to 5






Group: 1, type BRI: channels: 1 to 4

PIKA Channel Asterisk Group: 1, type BRI: channels: 1 to 4

Channel number: 1, asterisk group: 1 type: BRI, state = CHANNEL IS DOWN




Group: 3, type AUDIO: channels: 1 to 1

PIKA Channel Asterisk Group: 3, type AUDIO: channels: 1 to 1

Channel number: 1, asterisk group: 0 type: AUDIO, state = CHANNEL IS READY

If you want to use a specific channel of a BRI span, you can use PIKA/digital/number/$OUTNUM$ where number is

a value between one and eight, depending on the number of BRI spans on the appliance (each span has two channels).

Adding a GSM Trunk

Before adding a GSM trunk using FreePBX, you must have a GSM module installed on your appliance. To add a GSM

trunk, click the link Add GSM Trunk on the Add a Trunk Web page. The following screen appears.


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Select the GSM channel from the GSM Identifier menu. You must configure each GSM radio/SIM card combination

in the appliance individually. The Trunk Details text box shows the default configuration settings which should be

suitable in most cases. Change the values as required for your system configuration. Refer to GSM Configuration ( pg.

55) for information about the parameters. FreePBX stores the configuration information from the Trunk Details text

box in the file /etc/Asterisk/gsm_additional.conf file on the appliance. Optionally, assign values to any other fields as

needed. Position the mouse pointer over the field name for information about the field. Click Submit Changes when

you finish. To apply your changes, click Apply Configuration Changes at the top of the page.


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8 Adding Extensions

This section describes how to set up extensions in Asterisk for SIP, PIKA FXS, and PIKA line-out audio channels.

To add an extension in Asterisk, click the Setup tab at the top of the left navigation menu. In the left navigation menu,

click Extensions. The following Web page appears.

Adding a PIKA FXS Channel

On the Add an Extension Web page, in the Device list, click Other Custom Device and click the Submit button.

The following Web page appears.


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1. In the User Extension text box, type the number you want to use for the extension.

2. In the Display Name text box, type the name you want to associate with this extension.

3. In the dial text box, type PIKA/fxs/number, where number is the channel number of the FXS channel you want to

add. PIKA FXS channels are numbered from one to nine, depending on the modules you have installed on the

appliance. FXS channel one is always present, as it corresponds to the built-in FXS port.

4. Optionally, change other fields as needed. If you enable voice mail (disabled by default), you should set a voice mail

password.

• If you want the user to receive faxes by email, ensure that you click the Enabled box and type an email address in

the Fax Email text box.

5. Click Submit when you finish.


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6. To apply your changes, click Apply Configuration Changes at the top of the page.

Custom Caller ID Information for FXS Lines

It is not possible to configure caller ID for PIKA FXS lines through FreePBX because they are custom devices. You

must log on to the appliance and manually edit the PIKA configuration files. To open an SSH session, click the Tools

tab at the top of the left navigation menu and click JavaSSH. A new window opens. Enter the user name and password

at the prompts. The default user name is root and the password is pikapika.

To customize caller ID information for PIKA FXS lines, modify the entries in one of the following files, located in the

directory /etc/asterisk. There is one file for each of the possible number of FXS lines, with an entry for each FXS line.

Choose the file that corresponds to the number of FXS ports present in your system

• pika_fxs_1.conf - base board FXS port only, no FXS modules present

• pika_fxs_5.conf - base board FXS port plus one FXS module present

• pika_fxs_9.conf - base board FXS port plus two FXS modules present

The following example shows the default contents for the file pika_fxs_5.cfg. The example shows that the caller ID

value for FXS port 1 is FXS 1. Ensure that you change the line that corresponds to the User Extension and Display

Name you typed in the text boxes when you added the custom FXS extension.

callerid="FXS 1" <4001>

channels => 1


channels => 2


channels => 3


channels => 4


channels => 5

To activate the changes, you must restart Asterisk. Using the FreePBX GUI, click the Tools tab at the top of the left

navigation menu, click the Restart link in the left navigation menu, and click the button Restart Asterisk. When

Asterisk restarts, all active calls disconnect.

Adding a SIP Extension

On the Add an Extension Web page, in the Device list, click Generic SIP Device and click Submit. The following

Web page appears.


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1. In the User Extension text box, type the number you want to use for the extension.

2. In the Display Name text box, type the name you want to associate with this extension.

3. In the secret text box, type the password for the extension. You use this value when you configure the password for

your SIP phone.

4. Optionally, change other fields as needed. If you enable voice mail (disabled by default), you should set a voice mail

password.

• If you want the user to receive faxes by email, ensure that you click the Enabled box and type an email address in

the Fax Email text box.

5. Click Submit when you finish.

6. To apply your changes, click Apply Configuration Changes at the top of the page.

The FreePBX System Status page shows the number of SIP phones registered with Asterisk. If you want to verify the


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registration status of each SIP phone, type sip show peers at the Asterisk CLI (refer to Using the Asterisk CLI ( pg.

48) for instructions).

Adding a PIKA Line-out Channel

The appliance is equipped with a line-out port that you can use to connect the appliance to a public address system. To

add an extension for the port, follow the steps above for adding a PIKA FXS channel. In the dial text box, type

PIKA/audio/1.

To test the line-out extension, connect your public address system to the appliance. Plug in a phone to any FXS port

and dial the extension of the line-out port you added. When you speak, you should hear your voice on the public

address system.


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9 Conferencing

To add a conference, click Conference in the left navigation menu. The following Web page appears.

In the Conference Number text box, type the number you want to use for the conference. Type a descriptive name for

the conference in the Conference Name text box. Optionally, change any other settings as required. If you want to use

a recording to greet the conference participants, you must have at least one system recording. Click the Join Message

arrow and select a recording. Click the Submit Changes button and click Apply Configuration Changes at the top of

the page when you finish.

PBX users can dial the conference only from other PBX extensions. If you want to allow users to dial the conference

directly from outside the PBX, you must configure an inbound route. Refer to section Adding Routes ( pg. 26) for


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instructions. If you want to allow users to dial the conference from outside the PBX through an IVR, you must select it

as one of the destinations. Refer to Setting Up an IVR ( pg. 24) for instructions.


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10 Setting Up an Interactive Voice Response

System

You can use FreePBX to configure the appliance as an IVR (Interactive Voice Response) system. Visit

http://www.freepbx.org/support/documentation/administration-guide/creating-an-ivr for more details.

To create an IVR, you must first activate a system recording. Refer to section System Recordings ( pg. 50) for

instructions. To add an IVR, in the left navigation menu, click IVR. The following Web page appears.

Click the button Add IVR on the right side of the page. The following Web page appears.


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http://www.freepbx.org/support/documentation/administration-guide/creating-an-ivr

In the text box Change Name, type a descriptive name for the IVR. Click the Announcement arrow to select a

recording. Choose one of the recordings you added previously. Click the Save button in the middle of the page.

Changes to any of the fields above are saved.

If your PBX configuration includes a conference and you want to allow users outside the PBX to dial the conference,

type the conference extension in the text box under the Return to IVR check box and click Conference. When you

finish, click the Save button at the bottom of the page and click Apply Configuration Changes at the top of the page.


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11 Adding Routes

Before adding a route, you must configure one or more trunks to use with the route. Refer to section Adding Trunks (

pg. 12) for instructions.

Outbound Routes

To add an outbound route, click the link Outbound Route in the left navigation menu. The following Web page

appears.

To add a route, type a descriptive name in the Route Name box. Add one or more dial patterns either by clicking an

item in the Dial patterns wizards menu or by typing the information in the Dial Patterns text box. Select a trunk to

use for the route from the Trunk Sequence menu. Click Submit when you finish. To apply your changes, click

"Apply Configuration Changes" at the top of the page.


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FreePBX has a default route for dialing 9 to call a number outside the PBX. You can change the route to use PIKA

trunks. Click the button 0 9_outside on the Add Route page. The Edit Route page appears. Select a trunk to use for

the route from the Trunk Sequence menu. Click Submit when you finish. To apply your changes, click Apply

Configuration Changes at the top of the page.

Least Cost Routing

You can configure more than one trunk for an outbound route. This may be helpful, for example, if you have different

service providers and prefer to use one for local calling and another for long distance calls. When you configure your

local route, in the Trunk Sequence list, select the trunk you want to use for local calling. Click the Add button and

another list appears. Select another trunk from the list. Calls placed to numbers that match the dial patterns for the

route attempt to use the first trunk in the list but will use the next trunk in the list if the first is unavailable.

The following Web page shows an example of a local route that first attempts to use FXO trunk four and will use FXO

trunk two if the first is busy.


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Inbound Routes

Before you add an inbound route, you must add a destination, such as an IVR, an extension, a conference, or a fax

extension. If you want to configure access to a conference extension from outside the PBX, you must first configure

one or more conferences.

You can have inbound routes based on:

• number of the called party, also referred to as direct inward dial (DID)

• you use this if you want to allow calls to PBX users without the calling party dialing the extension

• caller ID of the calling party (CID)

• a default route which is not based on either DID or CID

You should have a default inbound route , usually with an IVR as the destination, for all calls that do not match any

other inbound routes. When you add the default route, do not type values in the DID or CID text boxes.


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Creating a DID for PIKA FXO lines

You can use a PIKA FXO trunk for a DID route, however, this functionality is not available through FreePBX because

PIKA channels are custom trunks. You can use FreePBX to add a DID route for BRI and SIP trunks because they have

the called number in the protocol messages. There is a separate FreePBX Web page to configure a DID route for GSM

trunks.

To configure PIKA FXO trunks for DID, you need to log on the appliance and manually edit configuration files. To

open an SSH session, click the Tools tab at the top of the left navigation menu and click JavaSSH. A new window

appears. Enter the user name and password at the prompts. The default user name is root and the password is pikapika.

The following example shows how to assign the DID 1234567 to FXO line 1 and the DID 5551234 to FXO line 2.

Add the following to the file /etc/asterisk/extensions_custom.conf. Do not edit the file

/etc/asterisk/extensions_additional.conf because FreePBX overwrites it each time configuration changes are applied.

FreePBX does not overwrite the file /etc/asterisk/extensions_custom.conf.

[macro-from-pika-fxo-1]

include => macro-from-pika-fxo-1-custom

exten => s,1,Noop(Entering macro-from-pika-fxo-1 with DID = ${DID} and setting to: 1234567)

exten => s,n,Set(__FROM_DID=1234567)

exten => s,n,Goto(from-trunk,1234567,1)

; end of [macro-from-pika-fxo-1]

[macro-from-pika-fxo-2]

include => macro-from-pika-fxo-2-custom

exten => s,1,Noop(Entering macro-from-pika-fxo-2 with DID = ${DID} and setting to: 5551234)

exten => s,n,Set(__FROM_DID=5551234)

exten => s,n,Goto(from-trunk,5551234,1)

; end of [macro-from-pika-fxo-2]

[from-pika-fxo]

include => from-pika-fxo-custom

exten => _X.,1,Set(DID=${EXTEN})

exten => _X.,n,Goto(s,1)

exten => s,1,Noop(Entering from-pika-fxo with DID == ${DID})

exten => s,n,Ringing()


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exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})})

exten => s,n,Noop(DID is now ${DID})

exten => s,n,Set(CHAN=${CHANNEL:4})

exten => s,n,Set(CHAN=${CUT(CHAN,/,3)})

exten => s,n,Macro(from-pika-fxo-${CHAN},${DID},1)

exten => s,n,Noop(Returned from Macro from-pika-fxo-${CHAN})

exten => s,n,Goto(from-pstn,${DID},1)

exten => fax,1,Goto(ext-fax,in_fax,1)

; end of [from-pika-fxo]

Edit the file /etc/asterisk/pika.conf file and change the context to from-pika-fxo instead of from-pstn as highlighted

below.

[fxo]

gp_group=0

conf_ref=TRUNK_23800186

context=from-pika-fxo

extension=s

If you want to create additional DID numbers for your other PIKA FXO lines, add macro-from-pika-fxo-number as

shown above where number is the channel number. Any channel that does not have a macro-from-pika-fxo-number

associated with it will use the from-pstn context to route the call.

You can now use FreePBX to create an incoming route based on DID using PIKA trunks. Click the link Inbound

Route in the left navigation menu. The following Web page appears.


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In the text box Description, type a descriptive name for the route. In the DID Number text box, type the the DID

number you added to /etc/asterisk/extensions_custom.conf. Click the destination to which you want to route the call

(IVR, specific extension, voice mail, etc.). Optionally, assign values to any other fields as needed. Position the mouse

pointer over the field name for information about the field. If you want the caller to hear ring back while the PBX

completes the connection to the destination, click the checkbox Signal RINGING. If the destination extension is a line

shared between a user and a fax machine, you may want to insert a pause before the destination answers. Type a number

in the text box Pause Before Answer which is the number of seconds the PBX waits before answering the call.

Click Submit when you finish. To apply your changes, click Apply Configuration Changes at the top of the page.

Creating an Incoming Caller ID Route

You can create an inbound route based on the incoming caller ID. An incoming route can have both a Caller ID


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Number and a DID Number. If you check CID Priority Route, the caller ID route takes precedence.

Adding a GSM Channel DID

You can configure a DID number for GSM trunks, which is useful if you want to do call routing based on the GSM

trunk used for an incoming call. Before you add a DID, you need to add a GSM trunk as described in section Adding

Trunks ( pg. 12). To add a GSM Channel DID, click the Setup tab at the top of the left navigation menu. In the left

navigation menu, click GSM Channel DIDs. The following Web page appears.

You need to supply values for all the fields. For information about each item, position the mouse pointer over the field

name.

1. In the Channel field, type the channel number of one of the GSM trunks you created previously. Valid values are

1,2,3, or 4, depending on whether you have one or two GSM modules installed in the appliance. Each GSM module

has two channels.

2. Set DID to the number you want to use, typically this is your cell phone number.

3. Click Submit Changes when you finish. To apply your changes, click Apply Configuration Changes at the top of

the page.


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12 System Settings

You can use the system administration pages to:

• view system information

• update language translations

• configure the appliance on your network

• set up an email server

• restart the appliance and restart Asterisk

• show channel status

To see the menu selections for the WARP appliance, click the Tools tab at the top of the left navigation menu. The

section WARP System Administration in the left navigation menu shows the links to the pages listed above.

Appliance Information

To view system information, click the Information link in the left navigation menu. The following Web page appears.


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The information under the General Information heading shows details about the appliance hardware. The information

under the Packages heading lists the packages installed on the system, including the package version numbers. This

information may be useful if you need to troubleshoot problems with the WARP appliance.

Language Translations

If you update any modules using the Module Admin ( pg. 44) page, you must refresh language translations, otherwise,

language translations do not work. To update language translation files, click the Languages link in the left navigation

menu. The following Web page appears.


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Click the Update button to start the update. When it finishes, the page shows the message Finished updating the

WARP i18n language translation files and the FreePBX System Status page appears.

Mail Server Configuration

If you want users to receive emails, for example, for voice mail or fax notifications, you must configure an email server.

To configure an email server on the appliance, click the Mail Server Config link in the left navigation menu. The

following Web page appears.

In the text box MAILHUB text box, type the name or IP address of your email server. In the HOSTNAME text box,

type the domain name you want to appear as the sender's email address for emails sent from the appliance. In the

example shown, emails sent from the appliance show the sender's email address as [email protected]. Optionally,

assign values to any other fields as needed. Position the mouse pointer over the field name for information about the

field. Click the Save button when you finish. The message Configuration Saved appears at the top of the panel. You

do not need to reload Asterisk to apply your changes.


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Network Configuration

To configure the appliance on your network, click the Network Config link in the left navigation menu. The following

Web page appears.

DHCP is enabled by default on the appliance. When you connect the appliance to your network, the DHCP server on

your network issues a valid IP address to the appliance during the startup sequence. If your network does not have a

DHCP server, the appliance has an IP address in the range 169.254.x.y. If you want to use a static IP address, you can

use the network configuration page to change the network settings. To use a static IP address, in the DHCP_LAN text

box, type no. In the IP_LAN text box, type the IP address you want to use for the appliance. Change the values for the

other fields as required for your network. You must enter values for all fields if DHCP_LAN is no.

Click the Save button. The message Configuration Saved. You must reboot the device for changes to take effect

appears at the top of the panel. Click Restart in the left navigation panel and click the button Restart WARP to restart

the appliance.

Restart

To restart either the appliance or Asterisk, click the Restart link in the left navigation menu. The following Web page

appears.


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To restart the appliance, click the button Restart WARP. The following message appears: Restarting the WARP,

please wait until page refreshes. This will take a few minutes. When the appliance finishes restarting, the FreePBX

System Status page appears.

To restart Asterisk, click the button Restart Asterisk. The following message appears: Restarting Asterisk, please

wait until page refreshes. When Asterisk finishes restarting, the FreePBX System Status page appears.

PIKA Channel Status

You can use FreePBX to check the status of PIKA channels. This can be useful for troubleshooting purposes or to

verify group numbers when you edit files for manual configuration. To view the status of all PIKA channels, click the

Show Channels link in the left navigation menu. A Web page such as the following appears.


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In the example, FXS channel four has an active call. The BRI channels are not configured.


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13 Fax

You can use the PIKA fax module to send and receive faxes in Asterisk. You can upload a file from your PC and send it

to an extension or use a trunk to send it to a destination outside the PBX.

Fax Configuration

To configure the appliance to send a fax using an outbound channel, click the Tools tab at the top of the left navigation

menu and click Fax Config in the left navigation menu. The following Web page appears.

In the Fax Channel text box, type the name of the outbound channel to use to send a fax using the same format as

described in section Adding Trunks ( pg. 12). Optionally, change the other values on the page. Click Save when you

finish. To apply your changes, click Apply Configuration Changes at the top of the page.

Sending a Fax

You can upload a file from your computer in either TIFF or PDF format and send it as a fax using Asterisk. Note that

you need the ghostscript package installed on the appliance to send a fax in PDF format. This package is installed by

default on the appliance. If you want to receive a notification message when the fax is sent, ensure that you configure a


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mail server. Refer to section System Settings ( pg. 33) for instructions.

You also need to edit the file /persistent1/var/www/html/admin/modules/fax/bin/fax-process.pl. Log on to the

appliance using an SSH client. To open an SSH session using FreePBX, click the Tools tab at the top of the left

navigation menu and click JavaSSH. A new window appears. Type the user name and password at the prompts. The

user name is root and the password is pikapika. Change the following line in the file:

my $smtp = Net::SMTP-> new("127.0.0.1", Debug => 0) or

Replace 127.0.0.1 with the IP address of the mail server you configured.

Before you send a fax to an extension, ensure that you enable fax and configure an email address for the extension.

Refer to Adding Extensions ( pg. 17) for more information. To send a fax to an extension on the appliance, click Fax

Drop in left navigation menu. The following Web page appears.

In the Fax Number text box, type the phone number of the fax machine to which to send the fax. If you want to send

a fax to an outside number, you do not need an outbound route, such as 9_outside because you configure the fax

channel to use from the Fax Config Web page. When you type the number, do not use an outbound prefix, such as 9.

Click Browse and select the file from your computer that you want to send. Click tiff or pdf to match the format of the

file to send. In the Email Notification text box, type the email address of the recipient of the fax status notification

message. When you finish, click Upload. The following message appears. Click OK to continue.


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The following Web page appears to confirm that Asterisk sent the fax.

To track the history of each fax sent, click Fax Status in the left navigation menu. The following page appears.


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Receiving a Fax

You can configure a user extension so that the user receives a email with the fax attached when a fax is sent to their

extension. When you add the user extension, ensure that you click the check box Enabled in the Fax section of the

page. Type the user's email address in the Fax Email text box.

If you connect a fax machine directly to an FXS port on the appliance, and you want to receive faxes from fax machines

outside the PBX, you need to configure an inbound route for the extension. On the Inbound Routes page, click Yes

beside the Detect Faxes option. Choose the destination Fax Recipient extension, where extension is the user extension

corresponding to the FXS port to which the fax machine is connected. The destination selected in the Set Destination

section of the Web page receives all non-fax calls.

To configure the PBX to receive incoming faxes, click Fax Configuration in the left navigation menu. The following

Web page appears.


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The options on this page configure the appearance and transmission settings for sending a fax. For information about

each field, position the mouse over the field name. When you finish, click Submit and Apply Configuration Changes

at the top of the page.


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14 Module Admin

You can upgrade individual FreePBX modules when new versions become available. The FreePBX Notices section on

the System Status page shows whether new modules are available and if there are upgrades for any existing modules.

NOTE: Do not upgrade to a new version of FreePBX using the Module Admin page. You will lose all PIKA

customizations. You need to install a new release of Warp for Asterisk/FreePBX, if available, that uses the

newer version of FreePBX.

To upgrade a module or add a new module, click Module Admin in the left navigation menu. The following Web page

appears.


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Click the link Check for updates online to update the state of each module. The Web page updates the status of each

module. Items shown in red are installed but have an upgrade available. Items shown in blue are not installed. Note that

not all modules are available online, for example, the PIKA Fax module.

To upgrade all installed modules, click the link Upgrade all on the right side of the page. To install, uninstall, or

upgrade individual modules, click the module name. A menu with a list of actions appears underneath the module name.

Click the action that you want to perform and click Process in right upper corner to accept the action. Click the button

Confirm on the next screen. An orange progress box appears to show download progress. Click the link Return when

complete. Click Apply Configuration Changes at the top of the page.


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15 Backup and Restore

You can back up and restore appliance files using FreePBX as part of routine maintenance. Visit

http://www.freepbx.org/support/documentation/module-documentation/backup-and-restore for details.

Click the Tools tab at the top of the left navigation menu and click Backup and Restore. The following Web page

appears.

We recommend that you include files in the directory /etc/pika. Click Additional Files. In the text box Additional

files and folders, type /etc/pika/*. FreePBX stores backup files on the SD card in the directory

/mnt/sd/var/lib/asterisk/backups.


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http://www.freepbx.org/support/documentation/module-documentation/backup-and-restore

We do not recommend that you use the Backup and Restore page when you upgrade the appliance software. You

should use the backup method described in the PADS User Guide section Upgrading Software Using an Autorun

Script. Upgrade the appliance software according to the instructions in the appliance Release Notes.


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16 Miscellaneous

16.1 Using the Asterisk CLI

To use the Asterisk CLI, you can use either the FreePBX GUI or you can log on to the appliance using an SSH

connection. To use the Asterisk CLI from the FreePBX GUI, click the Tools tab at the top of the left navigation menu.

In the left navigation menu, click Asterisk CLI. The following Web page appears.

Type the command you want to run in the Command text box and click the Execute button. The command output is

displayed on the Web page. For a list of commands, type help in the Command text box and click the Execute button.

16.2 SD Card Usage

FreePBX stores files on the SD Card to reduce the space used in either persistent storage or in memory. We strongly

recommend that you use an SD card. For compatibility with Asterisk, the SD card must use an ext2 or ext3 file system

format. FreePBX automatically formats the SD card and transfers the necessary files when the appliance starts. If the

transfer has already been done, it is not repeated.

If you do not want FreePBX to format your SD card:

• Ensure that your SD card uses an ext2 or ext3 file system format.

If you do not want FreePBX to move files to the SD card:

PIKA WARP the Appliance for Asterisk / 16.2 SD Card Usage

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• Create a folder called usr/src/ on the SD card (/mnt/sd/usr/src).

FreePBX moves following directories to the SD card if it is present:

• /var/lib/asterisk

• backup files created using the Backup and Restore Web page

• music on hold files

• /var/spool/asterisk

• voice mail files

• incoming SMS files

• /var/www/html/admin/modules/cache

• /tftpboot

• used auto-provisioning IP phones

If you want to move additional files to the SD card automatically when the appliance starts, you can change the function

file_transfer in the file /persistent/autorun/S20sdtransfer. The script does the following:

• checks if the SD card is installed

• checks if the files are already on the SD card

• if files are not on the SD card, it formats the SD card and transfers the files to it

A portion of default file is shown below. Note the comment that shows where to add your changes.

file_transfer () { # move asterisk spool directory, especially to give voicemail more room mkdir -p /mnt/sd/var/spool mv /persistent1/var/spool/asterisk /mnt/sd/var/spool/ ln -sf /mnt/sd/var/spool/asterisk /persistent1/var/spool/ ... # move the moh (music on hold) directory mkdir -p /mnt/sd/var/lib/asterisk mv /persistent1/var/lib/asterisk/moh /mnt/sd/var/lib/asterisk/ ln -sf /mnt/sd/var/lib/asterisk/moh /persistent1/var/lib/asterisk/ ln -sf /mnt/sd/var/lib/asterisk/moh /persistent1/var/lib/asterisk/mohmp3 # insert any additional file moves HERE # transfer FreePBX source to SD card to free room for MySQL # on that partition. This move should ALWAYS be last one in # this function if you expand it mkdir -p /mnt/sd/usr/src if [ $? -ne 0 ] ; then return 1 fi ...

16.3 System Recordings PIKA WARP the Appliance for Asterisk /

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16.3 System Recordings

You need a recording to use with an IVR system (refer to Setting Up an IVR ( pg. 24)).You can also use a recording

to greet conference participants (refer to Conferencing ( pg. 22)). You can:

• create your own recording using Asterisk and FreePBX

• upload an existing recording

• use one of the recordings provided with FreePBX and Asterisk

To add a system recording, click the Setup tab at the top of the left navigation menu. In the left navigation menu, click

System Recording. The following page appears.

You can add a new recording by uploading a file from your computer or recording a message directly using Asterisk.

The create your own recording, you must first add one or more extensions. Refer to section Adding Extensions ( pg.

17). Type the extension in the text box indicated on the Web page above and click Go. The following Web page appears.

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Use a standard phone set or SIP phone that corresponds to an extension you added previously, dial *77, and record

your message when you hear the tone. Hang up when you finish. Optionally, dial *99 to listen to the recording. In the

Name this Recording text box, type a descriptive name for the recording. When you finish, click Save.

You can also use one of the built-in system recordings. To use a built-in recording, click the button Built-in

Recordings on the right side of the Web page. The following page appears.

16.3 System Recordings PIKA WARP the Appliance for Asterisk /

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In the Select System Recording box, select one of the recordings. Click Go. The following page appears.

If you want to change the name or description, type the new name and description in the associated text box. Click Save

and click Apply Configuration Changes at the top of the page.

PIKA WARP the Appliance for Asterisk / 16.4 Customizing the FreePBX Logo

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16.4 Customizing the FreePBX Logo

You can change the appearance of FreePBX to use your own logo instead of the PIKA logo on the upper right corner

of the Web pages. Your logo image must be 55 pixels in height. To use your own logo:

1. Copy your logo file from your computer to /persistent1/var/www/html/admin/images. You can use scp:

• At the Linux command prompt on your computer, type scp yourlogo

root@IPAddress:/persistent1/var/www/html/admin/images, where yourlogo is the file name of your logo file

and IPAddress is the IP Address of the appliance

• Type the password when prompted, the default password for the root user is pikapika

2. Log on to the appliance.

• To open an SSH session, click the Tools tab at the top of the left navigation menu and click JavaSSH. A new

window opens.

• Type the user name and password at the prompts. The default user name is root and the password is pikapika.

3. In the file /persistent/etc/amportal.conf on the appliance, change the file name on the line shown below to the file

name of your logo.

# AMPADMINLOGO: Defines the logo that is to be displayed at the TOP RIGHT of the admin screen.

# This enables you to customize the look of the administration screen.

# NOTE: images need to be saved in the ..../admin/images directory of your AMP install

# This image should be 55px in height

AMPADMINLOGO=pika-logo.png

16.5 General Settings

The General Settings Web page allows you to configure global PBX settings. In the left navigation menu, click General

Settings. The following Web page appears.

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Note that the default setting for the option Check for Updates in the section Online Updates is Yes. If you do not

provide an email address to receive the update notices, an error appears on the on the System Status page. If you do not

want to receive notifications about updates, you should change this option to No. If you provide an email address,

ensure that you configure a mail server. Refer to section System Settings ( pg. 33) for instructions.

PIKA WARP the Appliance for Asterisk / 16.5 General Settings

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17 Channel Drivers

Channel drivers support the appliance hardware with Asterisk. PIKA provides two channel drivers:

• GSM channel driver - supports the GSM module

• PIKA channel driver - supports analog and BRI modules

Each channel driver has its own set of configuration files to customize the channel driver behaviour. There are Asterisk

CLI commands associated with each channel driver to display status information and to enable debugging. To use these

commands, you can use FreePBX to access the Asterisk CLI (refer to section Using the Asterisk CLI ( pg. 48)) , or

you can log on to the appliance using SSH (click JavaSSH in the left navigation menu) and access the Asterisk CLI by

typing asterisk -r at the Linux command line on the appliance.

17.1 GSM Channel Driver

17.1.1 GSM Configuration

There are two configuration files for the chan_gsm channel driver in Asterisk. You can find them in the /etc/asterisk

folder on the appliance. The files are called gsm.conf and gsm_additional.conf.

GSM Configuration

The file gsm.conf file contains settings that apply to all GSM channels in the system. The default file contents are as

follows.

[general] ;Directory where the sms files are stored.smsdir=/var/spool/asterisk/smsin ;if you want to periodically check for the homezonecheck_homezone=yes ;if you want to skip the '+' prefix, set this to yesskip_plus=no ;set this to yes to enable debugging via syslogsyslogdebug=no

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;set this to yes to enable radio/gsm optimzed dtmf detection mechanismrelaxdtmf=no ;set to yes in order to limit the double digit dtmf senddtmfbegin=yes ;all port configuration should go to gsm_additional.conf;which has a sample configuration that is commented out #include gsm_additional.conf

All options here are grouped under the section [general].

Parameter Name Description Valid Values Default Value

smsdir The directory where the SMS files are stored. any valid Linux

directory path

default

/var/spool/asterisk/smsin

check_homezone Indicates whether the GSM channel driver should

periodically check homezone (also measuring

signal strength).

If you type gsm show status at the Asterisk CLI,

the information in the Home Zone column

shows the result.

• yes

• no

yes

skip_plus Enables skipping the + prefix for a numbers that

use it. Use of a + prefix is service provider or

region specific.

• yes

• no

no

syslogdebug Enables debugging via SYSLOG. • yes

• no

no

relaxdtmf Enables radio/GSM optimized DTMF detection. • yes

• no

no

dtmfbegin Determines when DTMF clamping should be

done on the audio frames.• yes - at the

beginning of

frame

• no - at the end

of the frame

yes

The last line in the file includes the file gsm_additional.conf.

GSM Channel Configuration

The file gsm_additional.conf contains all the channel specific information for each GSM trunk on the appliance.

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FreePBX automatically creates this file when you add GSM trunks using the GUI. You must modify the GSM trunk

configuration using the FreePBX GUI (refer to the section Adding Trunks ( pg. 12)); if you edit the file directly,

FreePBX overwrites your changes the next time the appliance configuration changes.

;--------------------------------------------------------------------------------;; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;; this file must be done via the web gui. There are alternative files to make ;; custom modifications, details at: http://freepbx.org/configuration_files ;;--------------------------------------------------------------------------------;; [from-gsm-custom-1]sms_pdu_mode=nosmsc=resetinterval=0pin=noneinitfile=/etc/asterisk/gsm-init-filehidecallerid=noexten=scontext=from-gsmchannel=>1rxgain=0txgain=0

Parameter

Name

Description Valid

Values

Default Value

channel The channel number(s) the configuration applies to.

You can use multiple channels.

Comma

separated

list of

channel

numbers.

1

initfile The file that contains the collection of commands used

to initialize the GSM radio. If no value is assigned, a

default set of commands is used.

You should not change this setting unless instructed to

do so by PIKA Support.

A path to

an

initialization

file with the

correct

format.

/etc/asterisk/gsm-init-file

context The context to use in the dial plan for incoming calls.

We recommend that you do not change the default

context because GSM DID routes require it. If you

change it, ensure that you understand the impact.

A valid

context

name

from-gsm

exten The extension to use for incoming calls. A valid

extension

s

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pin The PIN (Personal Identification Number) your

service provider requires for SIM card authentication.

Set this option to none if your SIM card does not

require a PIN for operation. If you need to set a PIN,

ensure that you use the correct value for successful

authentication. Do not change the value if you do not

require a PIN.

• none

• Server

provider

specific

none

hidecallerid Specifies whether to transmit the caller ID information

for outgoing calls.• yes

• no

no

resetinterval Used internally by the channel driver, do not change. 0

smsc The SMSC (Short Message Service Center) number is

normally preset in your SIM card. If you do not know

the SMSC setting or do not need to change it, set this

option to none to use the value in your SIM card.

Using an incorrect value causes SMS messaging to fail.

• none

• Server

provider

specific

not set

sms_pdu_mode The format to use when receiving SMS messages: PDU

(protocol description unit) or text format. PDU format

includes non-ASCII characters.

• yes (use

PDU

format)

• no (use

text

format)

no

rxgain This value should normally remain 0 unless the voice

level is too low.

0

txgain This value should normally remain 0 unless the voice

level is too low.

0

17.1.2 Using the GSM CLI

The GSM channel driver has commands to do the following:

• channel commands

• SMS commands

• configuration

• debugging

• operator

Using the GSM CLI PIKA WARP the Appliance for Asterisk / 17.1 GSM Channel Driver

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Channel commands

The channel commands work on ports. The valid range for port in each command is 1 to 2 with one GSM module

installed or 1 to 4 with two GSM modules installed.

• gsm init port port

• Initializes a GSM port, where port is the number of the portto initialize.

• gsm shutdown port port

• Stops a GSM port, where port is the number of the port to stop.

• gsm restart port port

• Restarts a GSM port, where port is the number of the port to restart

• gsm show status

• Lists the status of all GSM ports. Sample output:

Port Status InUse Provider Home Zone Signal Quality

1 (0) REGISTERED n "Rogers Wireless" y -82dbm

2 (0) REGISTERED n "Rogers Wireless" y -67dbm

• Port - GSM port index

• Status - indicates if the port is connected to a GSM network, the values are REGISTERED or UNREGISTERED

• InUse - indicates if there is an active call on the GSM port

• Provider - the name of the GSM service provider.

• Home Zone - indicates if the GSM module is registered to the user's own GSM service provider

• Signal Quality - the GSM signal strength, the values indicate the following:

• not measurable (not measured yet)

• -112 dbm is very low

• -97 dbm is low

• -82 dbm is acceptable

• -67 dbm is good

• -52 dbm is high

• -51 dbm is very high

SMS Commands

• gsm send sms port number "smstext"

• Sends an SMS message where port is the GSM port to use on the appliance, number is the phone number of the

recipient and smstext is the message to send. For example, to send an SMS on port 1 to phone number 5556666,

type:

gsm send sms 1 5556666 "Hello"

• gsm show sms port

• Displays all stored SMS messages for the GSM port. For example, the response received for an SMS message on

port one:

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P(1) SMS RECEIVED: Idx(2) Number(+16133011736) Date(09/12/08,15:54:58+00) Text(testing 1,2,3,4)

• gsm fetch sms port

• Retrieves all stored SMS messages for the GSM port and saves them on the appliance in the directory

/var/spool/asterisk/smsin. Messages are erased from the GSM module SIM card memory. For example, for SMS

retrieved:

Saving SMS in /var/spool/asterisk/smsin/1260307679-254790.sms

Configuration Commands

• gsm show config values - displays all configuration values

• gsm show config helpall - displays all available configuration items

• gsm show config help parameter - displays the description for parameter

Debugging Commands

• gsm set debug level

• turns debug on or off, where level is 0 for debug off and 1 for debug on

• gsm debug info

• displays debugging information

• gsm show version

• displays the version information for the GSM channel driver

SMS Example

This example shows how to configure Asterisk to send an SMS message to phone number 6135558297 when any user

receives a voice mail.

Create a file called smsnotify.sh containing the following shell script code. Save it in the directory /var/lib/asterisk/bin

on the appliance. Ensure that the file has execute permissions and the owner is the asterisk user.

#!/bin/shtimestamp=`date`asterisk -r -x "gsm send sms 1 6135558297 \"New ($3) voicemail for extension $2 arrived on $timestamp\""

Edit the file /etc/asterisk/voicemail.conf on the appliance. Search for a line in the file with the text externnotify, for

example:

;externnotify=/usr/bin/myapp

Change the line to use the file you created:

externnotify=/var/lib/asterisk/bin/smsnotify.sh

To change the destination of the SMS message based on the extension associated with the voice mail box, you can

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change the script to check the parameter $2.

17.2 PIKA Channel Driver

The channel driver is a specialized piece of software that enables Asterisk to use the appliance FXO, FXS, and BRI

modules.

17.2.1 Configuration

When the appliance starts, the hardware discovery software generates the configuration files for the channel driver and

underlying PIKA software, based on the type of telephony modules installed on the appliance. Three configuration files

are generated:

• /etc/asterisk/pika.conf - channel driver configuration

• /etc/pika/pikagp.cfg - PIKA GrandPrix configuration

• /etc/pika/pikagp_aoh.cfg- PIKA GrandPrix configuration

In most cases, the default values are sufficient and it is not necessary to manually edit the configuration files. If you need

to edit the files, you must log on to the appliance using an SSH client. To open an SSH session using FreePBX, click the

Tools tab at the top of the left navigation menu and click JavaSSH. A new window appears. Type the user name and

password at the prompts. The user name is root and the password is pikapika.

Configuration files contain sections of associated key assignments, presented in an INI-file fashion. Sections are

denoted in the form [section name]. Key assignments are denoted by key=value.

This section describes the configuration settings for the channel driver. For details about PIKA GrandPrix

configuration files, refer to the PIKA GrandPrix Configuration Guide on the PIKA Support web page.

The configuration parameters are grouped in sections. The valid parameters are described below.

Global Parameters

The parameters in the [general] section of the file /etc/asterisk/pika.conf apply to all BRI, FXO and FXS channels.

usenativebridge

• Specifies whether to use native bridging for calls between PIKA channels. The voice path of the channels connect

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http://www.pikatechnologies.com/english/View.asp?x=344

directly without going through Asterisk. In most cases, the value should remain yes for the best performance and

audio quality. Default_value = yes.

echocancelwhenbridged

• Specifies if echo cancellation should be used on bridged calls. Default_value = yes.

busycount

• The number of busy tones to wait for before hanging up. A higher number will lower the probability of random

hangups, but will increase the time required to hangup a channel. The default is 2.

busypattern

• Specifies the cadence of the busy signal.The format is tone_length,silence_length. In many countries, it is 500 ms on, 500

ms off.

The following shows the default configuration entries for this section.

[general]

usenativebridge=yes

echocancelwhenbridged=yes

busycount=2

busypattern=500,500

The parameters in the section [audio] configure the appliance audio line-in and line-out ports.

txgain

• Changes the outgoing volume level on the channels. Valid values are -18 to 6. Default_value = 0

rxgain

• Changes the incoming volume level on the channels. Valid values are -18 to 6. Default_value = 0

mute

• Mutes the audio ports. Valid values are 0 (false) or 1 (true). Default value = 0

compandmode

• Audio format to use when connecting to other channels through Asterisk. Valid values are 0 (mulaw) and 1 (alaw).

Default value = 0

The following shows the default configuration entries for this section.

[audio]

rxgain=0

txgain=0

mute=0

compandmode=0

Group parameters

The following parameters are common across all channel types.

gp_group

• A reference between the pika.conf file and a configuration file used internally by the channel driver. It is not used to

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configure any channel driver behavior and should not be changed by the user.

conf_ref

• A reference between the pika.conf file and a configuration file used internally by the channel driver. It is not used to

configure any channel driver behavior and should not be changed by the user.

context

• The initial default context for the dial plan. Default value = from-internal for FXS lines and from-pstn for FXO

and BRI trunks

extension

• The initial default extension for the dial plan (used for FXO interfaces and FXS interfaces if immediate=yes is set).

For digital interfaces, leave this keyword empty to receive DID in the ${EXTEN} variable. Default value = s.

accountcode

• This field, if defined, is used to populate the accountcode field of the CDR. It can be any string up to 20 characters.

Default=fxs_grp for FXS lines, fxo_group for FXO trunks and digital_grp for BRI trunks

immediate

• Applies to FXS and BRI channels. Specifies whether to go immediately to the default context, exten right after the

FXS port goes offhook. Default_value = no.

mailbox

• If a mailbox is specified, a stutter dialtone will be heard when the phone is taken off-hook if the mailbox contains

new messages. By default, this keyword is not set.

echocancel

• Specifies whether to enable echo cancellation. Default value = yes.

echotaillength

• The tail length value, in milliseconds, when echo cancellation is enabled. The range is 1 ms - 128 ms. The default

value is 64 ms. This value is usually sufficient for connection between any channels on the appliance. This parameter

should not be changed unless required because increasing it can adversely affect voice quality due to extra CPU use.

echosuppression

• The echo suppressor is designed to eliminate any residual echo and noise that exists after echo cancellation has

removed the predicted echo using the echo path model and reference signal input. For human listener applications,

this keyword must remain enabled. Default value = yes.

comfortnoise

• Enable comfort noise insertion. Default value = yes.

echologging

• Enables per channel echo cancellation logging. This should only be used when debugging audio path issues as it can

affect system performance. A message is generated at the Asterisk CLI when echo cancellation is activated as a

reminder that logging is on. Log files are stored either in the local directory from which Asterisk was started or in

/tmp. The log file name formats are ec_ref-mmmmmmmm-nnnm.raw and ec_sin-mmmmmmmm-nnnm.raw where

mmmmmmmm is the media stream ID and nnnn is the file count. Default value = no

usecallerid

• Specifies whether to detect callerid on incoming calls for FXO ports and whether to generate callerid for FXS ports.

Default value = yes.


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faxdetection

• Enable fax detection. Default value = yes.

musiconhold

• The music on hold class associated with this group. Default value = default.

language

• Language for prompts purposes mostly. Default value = default

amaflags

• Sets the AMA (Automated Message Accounting) flags for billing. Valid values are: default, omit, billing and

documentation. Default value = default.

canpark

• Specifies whether to allow parking of active calls (check features.conf for the parking extension, the default is 700). In

order to unpark the call later, the parking context must be included within the default context (check features.conf)

also; the default context name is parkedcalls. Default value = no.

txgain/rxgain

• These keywords change the volume levels on the channels. Valid values are from -300 (-30 db) to +300 (+30db).

Default_value = 0.

clippingratio

• Specifies the amount of clipping that will performed to the RTP input signal to prevent overload distortion. The units

are dB relative to the maximum signal level that can be represented as a mu-law or a-law value. This value should not

normally be changed. - Default=-2

useagc

• Specifies whether AGC (Automatic Gain Control) should be used. This feature does not work during native bridging

(usenativebridge=yes). Default value = no.

agc.in.enable

• Specifies whether AGC should be used for the receive side of the call.

agc.out.enable

• Specifies whether AGC should be used for the transmit side of the call.

agc.out.attackRate

• Attack rate for the transmit side of the call in ms.

agc.out.decayRate

• Decay rate for the transmit side of the call in ms.

agc.in.attackRate

• Attack rate for the receive side of the call in ms.

agc.in.decayRate

• Decay rate for the receive side of the call in ms.

agc_in.targetLevel

• Target power level for the receive side of the call in dBm0.

agc_in.minGain

• Minimum attenuation level for the receive side of the call in 1/10 dB.

agc_in.maxGain


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• Maximum amplification level for the receive side of the call in 1/10 dB.

agc_in.speechLevel

• Threshold in dBm0 for speech detection for the receive side of the call.

agc_out.targetLevel

• Target power level for the transmit side of the call in dBm0.

agc_out.minGain

• Minimum attenuation level for the transmit side of the call in 1/10 dB.

agc_out.maxGain

• Maximum amplification level for the transmit side of the call in 1/10 dB.

agc_out.speechLevel

• Threshold in dBm0 for speech detection for the transmit side of the call.

group

• Value from 0 to 31. This group is used to indicate which group the Dial command should use. E.g.:

Dial(PIKA/FXO/g0/1234). There is no default value. The group number is unique for different interface types

(FXS, FXO, BRI). The following can be used to specify the search algorithm to use when selecting a free channel in

the group:





channels

• One or more channels included in this group. Channels must always be the last configuration statement for a group.

The following parameters are specific to FXS channels.

cancallforward

• Specifies whether call forwarding is enabled, enabled by *72 and disabled by *73, unless overridden by callforwardon

and/or callforwardoff. Default value = no.

callforwardon

• Customizes the number to dial to enable call forwarding. Default value = *72.

callforwardoff

• Customizes the number to dial to disable call forwarding. Default value = *73.

lcterm_fxs

• If enabled, the channel driver issues an LC Term signal (Loop Current Termination) to the line. This signal is

typically used to allow automated dialing equipment to go on-hook when the remote end has ended the call. Default

value = yes.

callwaiting

• Enables or disables call waiting. Default value = no.

calltransfer

• Enables or disables call transfer. Default value = yes.


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The following shows the default settings in section [fxs]. The settings in this section apply to all FXS channels on the

appliance. This section is always present because there is always at least one FXS channel on the appliance, the built-in

FXS channel. The include entry at the bottom of the section indicates which of the caller ID files Asterisk uses. Refer

to section Adding Extensions ( pg. 17) for information about caller ID files.

[fxs]

gp_group=0

conf_ref=PHONE_0

context=from-internal

extension=s

accountcode=fxs_grp

cancallforward=no

callforwardon=*72

callforwardoff=*73

lcterm_fxs=yes

callwaiting=no

calltransfer=yes

immediate=no

mailbox=

echocancel=yes

echotaillength=64

echosuppression=yes

comfortnoise=yes

echologging=no

usecallerid=yes

faxdetection=yes

musiconhold=default

language=default

amaflags=default

canpark=no

txgain=0

rxgain=0

clippingratio=-2

useagc=no

agc.in.enable=no

agc.out.enable=no

agc.in.targetLevel=-15.0

agc.out.targetLevel=-15.0

agc.in.minGain=-6

agc.out.minGain=-6

agc.in.maxGain=18

agc.out.maxGain=18

agc.in.attackRate=170

agc.out.attackRate=170


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agc.in.decayRate=750

agc.out.decayRate=750

agc.in.speechLevel=-36

agc.out.speechLevel=-36

group=0

#include pika_fxs_5.conf

The following parameters are specific to FXO channels.

waitfordialtone

• If enabled, the channel driver will wait for the dial tone before dialing. Default value = yes.

hookflashco

• Specifies whether to allow hook flash to be sent to the CO. Applicable only to Centrex lines and/or lines with call

waiting service. Default value = no.

flashstring

• String to use to invoke hook flash on the trunk. The default is ##. Asterisk processes only the first 2 characters.

callerid

• Default=asreceived

The following shows the default settings in section [fxo].

[fxo]

gp_group=0

conf_ref=TRUNK_0

context=from-pstn

extension=s

accountcode=fxo_grp

waitfordialtone=yes

hookflashco=no

flashstring=##

callerid=asreceived

echocancel=yes

echotaillength=64

echosuppression=yes

comfortnoise=yes

echologging=no

usecallerid=yes

faxdetection=yes

musiconhold=default

language=default

amaflags=default

canpark=no

txgain=0

rxgain=0


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clippingratio=-2

useagc=no

agc.in.enable=no

agc.out.enable=no



agc.in.minGain=-6

agc.out.minGain=-6

agc.in.maxGain=18

agc.out.maxGain=18







group=0

channels=1-4

The following parameters are specific to BRI channels

callerid

• Default=asreceived

nationalprefix

• Default=0

internationalprefix

• Default=00

The following shows the default settings in section [digital]. Each span has its own set of configuration parameters.

[digital]

gp_group=1

conf_ref=BRI_1_0

briphone=false

context=from-pstn

extension=

accountcode=digital_grp

callerid=asreceived

nationalprefix=0

internationalprefix=00

immediate=no

echocancel=yes

echotaillength=64

echosuppression=yes


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comfortnoise=yes

echologging=no

usecallerid=yes

faxdetection=yes

musiconhold=default

language=default

amaflags=default

canpark=no

txgain=0

rxgain=0

clippingratio=-2

useagc=no

agc.in.enable=no

agc.out.enable=no



agc.in.minGain=-6

agc.out.minGain=-6

agc.in.maxGain=18

agc.out.maxGain=18







group=1

channels=1-2

conf_ref=BRI_1_1

briphone=false

context=from-pstn

extension=

accountcode=digital_grp

callerid=asreceived

nationalprefix=0

internationalprefix=00

immediate=no

echocancel=yes

echotaillength=64

echosuppression=yes

comfortnoise=yes

echologging=no

usecallerid=yes


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faxdetection=yes

musiconhold=default

language=default

amaflags=default

canpark=no

txgain=0

rxgain=0

clippingratio=-2

useagc=no

agc.in.enable=no

agc.out.enable=no



agc.in.minGain=-6

agc.out.minGain=-6

agc.in.maxGain=18

agc.out.maxGain=18







group=2

channels=3-4

Example

The following shows configuration examples for the appliance.

Configuring the Appliance for BRI

If there are one or more BRI modules present, the file contains configuration entries for the BRI interfaces. Depending

on your network configuration, you may need to change some settings. By default, each BRI span is configured as node

type client (TE) and networking configuration point-to-multi-point (PMP). PMP was chosen as the default for ease of

compatibility. For example, the terminal endpoint identifier (TEI) does not need to be pre-configured in a PMP

configuration.

In the configuration file /etc/pika/pikagp_aoh.cfg, there is a section entry for a BRI board. Regardless of the number

of BRI modules present (one or two), there is a single entry plus associated keys for the board. There is also a group

entry plus associated keys for each span. The file contains two or four span sections, depending on the number of

modules present.


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The following example shows BRI board group [DIGITAL_1]. The only key you may need to change is clockmode. In

most cases, you should use the default value, slave. This means that timing for the board is derived from the network.

The setting master applies only in loop back mode, which is typically used for testing.

[DIGITAL_1]

id=1

;serial=25800717

interfacetype=bri_eu

clockmode=slave

Examples of BRI span groups are shown further below. The important keys for BRI span groups are:

• networkconfig

• specifies whether the appliance is in a point-to-point or point-to-multi-point configuration

• values:

• pp - indicates point-to-point

• pmp - default value, indicates point-to-multi-point

• node

• specifies whether this span on the appliance is functioning as a TE or NT device

• values:

• client - default value, indicates a TE device

• network - indicates an NT device

Overlap Sending and Receiving

The appliance supports both overlap receiving and overlap sending.

Overlap receiving means that the Setup message for incoming calls does not contain the complete set of digits required

to route the call. Subsequent Info messages contain the remaining digits. Overlap receiving is enabled for the appliance

and may not be disabled. The appliance transparently handles both the enbloc or overlap methods for receiving digits.

Overlap sending means that the Setup message for an outgoing call does not contain all the digits necessary to route the

call. Info messages contain digits and the final Info message contains a "sending complete" flag to indicate that all the

digits have been provided. Overlap sending is disabled by default. To enable it, set the values as required for the

following keys on the appropriate span in /etc/pika/pikagp_aoh.cfg.

• addsendingcomplete

• specifies whether to add a sending complete information element to all outgoing setup messages

• values:

• yes - default value, all digits will be sent in the Setup message, indicates enbloc digit collection


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• no - digits may be sent in Info messages after the Setup message, enables overlap sending

• maxdigitsconnrq

• indicates the maximum number of called digits that may be placed in the outgoing Setup message.

• values:

• 0 - default value, all digits will be sent in the Setup message, used for enbloc digit collection

• a numeric value between 1 and 32 - Specifies the number of digits to send in the Setup message, subsequent

digits will be sent in an Info message, used with overlap sending enabled

Depending on the Asterisk dial plan used, you may need to change the value of the following parameter in the

/etc/asterisk/pika.conf.

• immediate

• indicates whether to start the dial plan before digit collection is complete

• values

• yes - start the dial plan and send the digits directly to Asterisk as DTMF digits

• no - the PIKA channel driver will attempt to match the digits with a existing extension before starting the dial

plan

The following example shows the settings for the first span on a board. It is configured as a TE device in a

point-to-point configuration. Overlap sending is disabled.

[BRI_1_0]

interface=board1,0

networkconfig=pp

switchtype=etsi

countrycode=europe

line_compandmode=alaw

node=client

numbertype=unknown

numberplan=unknown

channelselect=exclusive

endpoint=true

autoreconnect=false

addsotoend=false

resetchidextensionbit=false

enablet309=false

permanentactivation=false

dropteiondeact=false

;default_tei=0

;num_digits=0

;local_number=

;servicetype=voice

;in_calls_behavior=TRANSPARENT_OVERLAP_RCV

;out_calls_behavior=NONE


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;general_calls_behavior=CC_UUI_RECOGNITION

;addsendingcomplete=yes

;maxdigitsconnrq=0

;taillength=64

The following example shows the settings for the second span on a board. It is configured as an NT in a

point-to-multi-point configuration. Overlap sending is enabled.

[BRI_1_1]

interface=board1,1

networkconfig=pmp

switchtype=etsi

countrycode=europe

line_compandmode=alaw

node=network

numbertype=unknown

numberplan=unknown

channelselect=exclusive

endpoint=true

autoreconnect=false

addsotoend=false

resetchidextensionbit=false

enablet309=false

permanentactivation=false

dropteiondeact=false

;default_tei=0

;num_digits=0

;local_number=

;servicetype=voice

;in_calls_behavior=TRANSPARENT_OVERLAP_RCV

;out_calls_behavior=NONE

;general_calls_behavior=CC_UUI_RECOGNITION

addsendingcomplete=no

maxdigitsconnrq=1

;taillength=64

You must restart Asterisk for the configuration changes to take effect. Using the FreePBX GUI, click the Tools tab at

the top of the left navigation menu, click the Restart link in the left navigation menu, and click the button Restart

Asterisk.


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Configuring Asterisk on the Appliance for use Outside North America

The Asterisk channel driver on the appliance uses a North America configuration unless there is a BRI module installed,

in which case it uses a European configuration. You can use the application pikacf to change the configuration. After

you finish using pikacf, you must restart the appliance.

To configure the appliance for other countries, perform the following steps:

1. Type one of the following commands at the Linux command prompt on the appliance to stop Asterisk:

• if FreePBX is installed, type amportal stop

• if FreePBX is not installed, type asterisk -rx "stop now"

2. Run the configuration program to generate new configuration files. At the Linux command prompt on the appliance,

type pikacf.

• Choose the appropriate country code. If your country code does not appear, press c when presented with the

following prompt.

NOTE: BRI is not supported for North America, therefore, even if you select the country code US, settings chosen for

BRI reflect a European configuration.

Enter two letter country code, for example 'US' for United States.

(Enter '?' to display the list of supported country codes or

'c' to customize individual parameters)

Country code: [US]

• Choose the appropriate companding mode and caller id mode values for your country.

Compand mode, ulaw or alaw: (Enter '?' for help) [ulaw] alaw

Select one of the following caller id modes

1. FSK BELLCORE

2. FSK ETSI

3. FSK BT

4. FSK NTT

5. DTMF ETSI

6. DTMF DK

7. DTMF SE FI

8. None

Selection: (Enter '?' for help) [8]

After the configuration program completes, the configuration files contain the entries for any FXO, FXS and BRI ports

on the appliance.


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NOTE: Any country for which there is a predefined configuration will have the correct line impedance value set.

Configuration for other countries will use the North American value. There is no configuration file entry to

change the line impedance.

3. Change the indication tones for Asterisk in the file /etc/asterisk/indications.conf

• In the section [general], if your country is in the list of the default countries, modify the line "country=" as

appropriate.

• If your country is not in the list, use the template shown in the file to create a section that defines the tones for

your country.

4. Restart the appliance:

• At the Linux command prompt on the appliance, type reboot.

Call Progress

Call progress uses inband tone patterns (such as ringing, busy or fast busy) received from remote devices to determine

why a call has been disconnected. Enabling call progress ensures that billing does not start until a two-way speech

connection has been established. Call progress is enabled by default for the analog ports on the appliance. Only North

American call progress tones have been defined and therefore, call progress may not work outside of North America.

Two options are available for this situation.

1. Disable call progress (recommended)

• In the file /etc/pika/pikagp.cfg, change all entries in the file that match callpa=callpa_settings to

;callpa=callpa_settings.

• In the file /etc/pika/pikagp_aoh.cfg, change all entries in the file matching ;answer=speech or answer=speech

to answer=none.

2. Modify the tone definitions in /etc/pika/inccpa.cfg to correspond to the values for your country. The comments in

the file and the following websites provide information about defining the tones.

• www.3amsystems.com/wireline/tone-search.htm

• www.itu.int/dms_pub/itu-t/oth/02/06/T02060000040002PDFE.pdf

17.2.2 Using the PIKA CLI

The following lists the CLI commands for the PIKA channel driver.

pika debug {on| off}

17.2 PIKA Channel Driver PIKA WARP the Appliance for Asterisk / Using the PIKA CLI

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http://www.3amsystems.com/wireline/tone-search.htm

http://www.itu.int/dms_pub/itu-t/oth/02/06/T02060000040002PDFE.pdf

• Enables or disables debug messages

pika rec {on | off} {fromasterisk | toasterisk} gpgroup channel [nocli]

• Enables or disables audio recording for debugging purposes, either from Asterisk to the card (fromasterisk) or to

Asterisk from the card (toasterisk) where gpgroup is the GrandPrix group number and channel is the channel or

channel range to record. The option nocli disables CLI messages. We recommend that you do not use nocli.

pika reload

• Reloads the PIKA channel driver to activate configuration changes.

pika show audio

• Shows debugging information for the line-in and line-out ports. Used for internal PIKA debugging only.

pika show channels [gpgroup] [channel]

• Shows the status of the PIKA channels where gpgroup is the GrandPrix group number and channel is one or more

channels in gpgroup.

pika show info

• Shows the PIKA board and driver information

pika show version

• Shows the PIKA channel driver version information

17.3 Channel Status on the LCD

The LCD displays the status for each channel when Asterisk runs. The display reflects only PIKA channels, not other

Asterisk channels types, such as SIP channels.

During the startup sequence, the LCD shows the message WARP appliance...loading. After approximately two

minutes, the startup sequence finishes. When Asterisk runs on the system, the LCD on the appliance shows system

information and the line status of the channels. The LCD shows the IP address and when Asterisk starts, changes to

show the channel states. A single press of the LCD toggle button (beside the LED and highlighted below) switches the

display to show the IP address. Press the button again to return to the channel state display. The icons on the display

reflect different channel states.

FXS Channels

Icon Description

off-hook

on-hook or disconnected

ringing - alternates between on-hook and off-hook icons

PIKA WARP the Appliance for Asterisk / 17.3 Channel Status on the LCD

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FXO Channels

Icon Description

channel idle

channel in use

channel disconnected

BRI Channels and Spans

Note that the icons for BRI represent spans, not channels.

Icon Description

both channels on the span are idle

one or both channels on the span are in use

span disconnected, both channels are unavailable

GSM Channels

channel idle

channel in use

channel disconnected, the radio is not registered with the network

The following example shows the display where channel one (FXS) has a call in progress. A channel on BRI span two

and a channel on BRI span four have calls in progress. BRI spans three and five are active but currently idle. BRI spans

six through nine are not connected.

The following example shows the display where channel one (FXS), three (FXS) and eight (FXO) have calls in progress.

Channel nine (FXO) is active but is currently idle. Channels six and seven (FXO) are not connected.

17.3 Channel Status on the LCD PIKA WARP the Appliance for Asterisk /

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The following example shows the display where channel one (FXS) is idle, channel two (GSM) has a call in progress,

and channel three (GSM) is disconnected (no SIM card plugged in).

Port Numbering - Provisioning versus LCD

On the LCD, channels are numbered sequentially, one through nine, regardless of the module type inserted into the

module bay. When phones and trunks are provisioned in Asterisk configuration files, channels are numbered

sequentially based on the module type. FXS channels are always numbered one through five, where channel one is

always the built-in FXS port. FXO channels are always numbered one through four. BRI channels are numbered one

through eight in the dial plan (note that the LCD shows spans, not channels).

The following diagram shows an example of this mapping. The FXO module is installed in the first bay, which is closest

to the built-in FXS channel. The FXS module is in bay two. When you configure Asterisk (e.g. fxs/2 in the dial plan),

use the numbers as shown on the back view below. The bottom view of the LCD shows the same module installation

and the corresponding channel numbering.

PIKA WARP the Appliance for Asterisk / 17.3 Channel Status on the LCD

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If the appliance is installed such that the display is upside down, press the toggle button on the LCD face plate three

times in succession to reverse the display.

17.3.1 BRI Link State Display

Due to limitations of the BRI hardware and the link protocol, the LCD may not accurately represent the channel state

under certain conditions. This section attempts to summarize the expected behaviour. A span or channel is considered

connected when both ends of the link are connected within the telephony network and exchanging low-level signaling

information. The LCD reflects the state of the channel in Asterisk.

Appliance Startup

• If a span is connected when Asterisk starts, the LCD shows that the channels are connected and idle.

17.3 Channel Status on the LCD PIKA WARP the Appliance for Asterisk / BRI Link State Display

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• If a span is not connected, the LCD shows the empty icon box, which indicates that the channels are not connected.

When the span is connected, Asterisk updates the display to show that the channels are idle.

Call Processing

• When there is a call in progress, the display shows that the channel is in use.

• If the span is disconnected (the cable is unplugged either locally or at the far-end) during a call, Asterisk updates the

LCD immediately to show the empty icon box. When the channel is reconnected, Asterisk updates the LCD

immediately to reflect the change to the idle state.

• If the LCD shows that a channel is connected and Asterisk attempts a call on that channel, if the channel is actually

disconnected, the LCD will first show the channel in use, and then after a few seconds Asterisk updates the display to

show that the channel is disconnected.

Power Saving

The network side (NT) of a BRI span initiates power saving mode at a pre-determined time (usually ninety seconds)

following the completion of the last call on the span. Low-level signaling information between the NT and TE stops

and the channels on the span are deactivated. Note that the appliance cannot initiate power saving, even in NT mode. If

Asterisk attempts a call on a deactivated channel, the channel automatically exits power saving mode, the call progresses

normally, and the LCD accurately reflects the line state.

If a channel is considered connected when power saving mode is initiated, the LCD indicates that the channel is

connected even if it is subsequently disconnected (the cable is unplugged either locally or at the far-end). If Asterisk

then attempts a call on a disconnected channel, the call will fail and Asterisk update the LCD to indicate that the

channel is disconnected.

BRI Link State Display PIKA WARP the Appliance for Asterisk / 17.3 Channel Status on the LCD

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18 Troubleshooting

You can find additional information in the FAQ and Troubleshooting sections on the PIKA Technologies web site

http://www.pikatechnologies.com/appliancedownloads and in the Troubleshooting section of the PIKA Warp

the Appliance User Guide.

Voicemails are distorted when sent to /var/spool/asterisk

If this occurs within the number of simultaneous calls specified in performance benchmarks, use the FreePBX GUI

to configure Asterisk to send voice mails to an SD card.

BRI spans stay ready after unplugging a cable

Once a span has gone into power saving mode, which is usually initiated by the network side, the appliance hardware

does not detect a change in the link state. When an incoming or outgoing call is attempted, the state will be updated

accordingly.

BRI Module is not shown on the LCD or in Asterisk

Verify that the FPGA version is 2022 or later. The display on the serial console during the startup sequence shows the

current version. BRI modules do not work with older FPGA versions. To verify the FPGA version, you must first log

on to the appliance. To open an SSH session, click the Tools tab at the top of the left navigation menu and click

JavaSSH. A new window opens. Enter the user name and password at the prompts. The default user name is root and

the password is pikapika. At the Linux command prompt, type cat /proc/driver/taco | cut -c10-13 at the Linux

command prompt on the appliance. The return value is the FPGA version number.


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http://www.pikatechnologies.com/appliancedownloads

19 FreePBX FAQ

This section provides answers to some commonly asked FreePBX questions.

To change some configuration settings, you need to log on to the appliance. To open an SSH session, click the Tools

tab at the top of the left navigation menu and click JavaSSH. A new window opens. Enter the user name and password

at the prompts. The user name is root and the password is pikapika.

How can I change the email notification parameter?

Log on to the appliance and edit the file /etc/asterisk/voicemail.conf. Search for a line in the file with the text

serveremail and change the line to:

serveremail=gaytes

How can I access the MySQL database from a PC running Windows?

You need to change the database access permissions.

1. Log on to the appliance and type mysql -p at the Linux command prompt.

2. When the password prompt appears, type amp109.

3. At the mysql prompt type grant all privileges on *.* to 'asterisk'@'%' identified by 'amp109' with GRANT

option;

On your PC running Windows, use an SQL client to connect to the appliance using the credentials from the file

/etc/asterisk/cdr_mysql.conf:

dbname=asteriskcdrdb

user = asterisk

password = amp109

port=3306

How do I create extension, agent, and trunk groups?

For Agent-like functionality, install the Queues and Queue Priorities modules from the Module Admin page. Visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=queues and

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=queueprio for

configuration instructions. You can also install the Ring Groups module. Visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=ringgroups for

configuration instructions.


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http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=queues

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=queueprio

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=ringgroups

How can I set group policies and rules for features such as ACD, UCD, Hunting ?

For FXO and BRI group hunting, refer to section Adding Trunks ( pg. 12).

How can I get flexible agent routing policies?

• Most Idle

• Minimum Service Count

• Minimum Service Time

• Round Robin

• Skill Level Routing





How can I provide call queueing?





How do I create speed dial numbers such as emergency calls or directory assistance?

Install the Speed Dial Functions module from the Module Admin page and visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=speeddial for


Is it possible to play IVR greeting messages based on a specific date or time ?

Install the Announcements module from the Module Admin page and visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=announcement for

configuration instructions. You can use the Time Condition module for specific date and time events, visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=timeconditions for

configuration instructions. See also: http://www.freepbx.org/news/2008-09-25/time-groups-time-conditions

How do I set a wakeup alarm and play a message?

Install one of these third-party modules:

• http://pbxinaflash.com/forum/showthread.php?t=3491 which refers to

http://projects.colsolgrp.net/projects/list_files/hotelwakeup

• http://mirror.freepbx.org/modules/release/contributed_modules/wakeup-1.0.tgz


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http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=speeddial

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=announcement

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=timeconditions

http://www.freepbx.org/news/2008-09-25/time-groups-time-conditions

http://pbxinaflash.com/forum/showthread.php?t=3491

http://projects.colsolgrp.net/projects/list_files/hotelwakeup

http://mirror.freepbx.org/modules/release/contributed_modules/wakeup-1.0.tgz

How do I configure the appliance to enable users to retrieve voice mail messages from outside the PBX?

Install the Misc Destinations module from the Module Admin page and visit

http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=miscdests for


What is the maximum number of conference call participants?

It depends on channel types used to connect the participants to the conference. Refer to the Performance

Benchmarks section of the Release Notes.

How do I set do not disturb ?

Use the ModuleAdmin page to install the Do-Not-Disturb Module (DND).

How do I set call forward from an extension line?

Use the ModuleAdmin page to install the Call Forward Module.


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http://www.freepbx.org/freepbx-help-system?freepbx_version=2.5.2.1&freepbx_module=miscdests

Index

A

Adding Extensions 17

Adding Routes 26

Adding Trunks 12

B

Backup and Restore 46

BRI Link State Display 79

C

Changing the Default Passwords 9

Channel Drivers 55

Channel Status on the LCD 76

Conferencing 22

Configuration 61

Contacting PIKA Technologies 2

Copyright Information 1

Customizing the FreePBX Logo 53

F

Fax 39

FreePBX FAQ 82

G

General Settings 53

GSM Channel Driver 55

GSM Configuration 55

I

Introduction 3

L

Logging on to FreePBX 8

M

Miscellaneous 48

Module Admin 44

O

Overview 5

P

PIKA Channel Driver 61

R

Related Documents 4

S

SD Card Usage 48

Setting Up an Interactive Voice Response System 24

System Recordings 50

System Settings 33

T

Troubleshooting 81

U

Using FreePBX 6

Using the Asterisk CLI 48

Using the GSM CLI 58

Using the PIKA CLI 75


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