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Telephony System Integrator's Guide (Asterisk) - Citrix Knowledge ...

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<strong>Citrix</strong> EasyCall Gateway<br />

<strong>Telephony</strong> <strong>System</strong> Integrator’s <strong>Guide</strong><br />

for <strong>Asterisk</strong>-Based <strong>Telephony</strong> <strong>System</strong>s<br />

<strong>Citrix</strong> ® EasyCall Gateway 3.0


Copyright and Trademark Notice<br />

Use of the product documented in this guide is subject to your prior acceptance of the End User License Agreement. A printable<br />

copy of the End User License Agreement is included on your product media and in the documentation download page of the<br />

administration tool.<br />

Information in this document is subject to change without notice. Companies, names, and data used in examples herein are<br />

fictitious unless otherwise noted. No part of this document may be reproduced or transmitted in any form or by any means,<br />

electronic or mechanical, for any purpose, without the express written permission of <strong>Citrix</strong> <strong>System</strong>s, Inc.<br />

© 2006-2009 <strong>Citrix</strong> <strong>System</strong>s, Inc. All rights reserved.<br />

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trademarks or trademarks of <strong>Citrix</strong> <strong>System</strong>s, Inc. in the United States and/or other countries.<br />

RSA Encryption © 1996-1997 RSA Security Inc., All Rights Reserved.<br />

This product includes software developed by The Apache Software Foundation (http://www.apache.org/)<br />

Licensing: Portions of this documentation that relate to Globetrotter, Macrovision, and FLEXlm are copyright © 2005<br />

Macrovision Corporation. All rights reserved.<br />

This product includes open source PostgresSQL, released under the BSD license.<br />

Trademark Acknowledgements<br />

Adobe, Acrobat, and PostScript are trademarks or registered trademarks of Adobe <strong>System</strong>s Incorporated in the U.S. and/or<br />

other countries.<br />

<strong>Asterisk</strong> is a registered trademark of Digium, Inc.<br />

FreePBX is a registered trademark of Atengo, LLC.<br />

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Solaris is a registered trademark of Sun Microsystems, Inc. Sun Microsystems, Inc has not tested or approved this product.<br />

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All other trademarks and registered trademarks are the property of their respective owners.<br />

Document Code: July 24, 2009 (KP)


CONTENTS<br />

Contents<br />

Chapter 1<br />

<strong>Asterisk</strong> SIP Trunk Integration<br />

Equipment Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2<br />

<strong>Asterisk</strong>-Based <strong>Telephony</strong> <strong>System</strong> Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2<br />

Integrating with <strong>Asterisk</strong>. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3<br />

Integrating with trixbox . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4<br />

Verifying the Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4<br />

Call Detail Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4


iv<br />

<strong>Telephony</strong> <strong>System</strong> Integrator’s <strong>Guide</strong>


CHAPTER 1<br />

<strong>Asterisk</strong> SIP Trunk Integration<br />

This chapter describes how to integrate <strong>Asterisk</strong> ® -based telephony systems with the<br />

EasyCall Gateway through a SIP trunk. The EasyCall Gateway can be integrated with<br />

<strong>Asterisk</strong> 1.2 and 1.4 servers.<br />

Note <strong>Asterisk</strong> is an open source/free software implementation of a telephone private<br />

branch exchange (PBX). Like any telephony system, it allows a number of<br />

attached telephones to make calls to one another, and to connect to other<br />

telephone services including the public switched telephone network (PSTN).<br />

The integration described in this chapter can be done in tandem with EasyCall<br />

Gateway installation and configuration, which includes defining the properties of the<br />

telephony system trunk and defining any changes the EasyCall Gateway makes to<br />

telephone numbers before sending them to the telephony system for dialing. The<br />

EasyCall Gateway always listens on port 5060. Be default, the EasyCall Gateway<br />

communicates with the SIP trunk over port 5060; that port is configurable. For<br />

information about EasyCall Gateway configuration, refer to the EasyCall Gateway<br />

Administrator’s <strong>Guide</strong>.<br />

Those responsible for configuring the telephone system must have the specific<br />

characteristics for the site available before attempting to integrate the EasyCall<br />

Gateway. Integrating the telephone system with the EasyCall Gateway is similar to<br />

integrating the telephone system with a voicemail system.<br />

The following topics describe a sample configuration of <strong>Asterisk</strong>-based telephony<br />

systems:<br />

“Equipment Requirements” on page 2<br />

“<strong>Asterisk</strong>-Based <strong>Telephony</strong> <strong>System</strong> Integration” on page 2<br />

“Verifying the Integration” on page 4<br />

“Call Detail Records” on page 4


2 EasyCall Gateway <strong>Telephony</strong> <strong>System</strong> Integrator’s <strong>Guide</strong><br />

Equipment Requirements<br />

Only the EasyCall Gateway appliance is required to integrate with an <strong>Asterisk</strong>-based<br />

telephony system. The following diagram shows the integrated components.<br />

<strong>Asterisk</strong>-based telephony system integrated with the EasyCall Gateway<br />

As shown in the diagram, the SIP trunk makes telephone calls over an Internet<br />

connection. The use of SIP trunks does not require modification to your existing<br />

telephony system or Internet connections and does not require PRI hardware or<br />

telephony system upgrades. Only outbound SIP trunks are required; the number of<br />

trunk channels used can be scaled to your needs. For information on capacity<br />

planning, refer to the EasyCall Gateway Administrator’s <strong>Guide</strong>.<br />

<strong>Asterisk</strong>-Based <strong>Telephony</strong> <strong>System</strong> Integration<br />

This section contains configuration examples for <strong>Asterisk</strong>-based telephony systems.<br />

Use the first example as a guide if your <strong>Asterisk</strong> implementation does not have a<br />

graphical user interface. Many organizations use telephony system products, such as<br />

FreePBX and trixbox, which extend the telephony features of <strong>Asterisk</strong> and include a<br />

graphical user interface. The trixbox example is provided as a guide for such systems.<br />

“Integrating with <strong>Asterisk</strong>” on page 3<br />

“Integrating with trixbox” on page 4


Chapter 1 <strong>Asterisk</strong> SIP Trunk Integration 3<br />

Integrating with <strong>Asterisk</strong><br />

For <strong>Asterisk</strong>-based telephony systems that do not have a graphical user interface, you<br />

must modify two configuration files and run two commands from the <strong>Asterisk</strong><br />

command line interface, as follows.<br />

To integrate with <strong>Asterisk</strong>-based telephony systems<br />

1. Modify the sip.conf file so that it contains the following:<br />

[general]<br />

context=default<br />

bindport=5060<br />

[EasyCallTrunk]<br />

context=easycall<br />

type=friend<br />

host=1.2.3.4 ; IP address of EasyCall Gateway<br />

port=5060<br />

disallow=all<br />

allow=ulaw<br />

dtmfmode=rfc2833<br />

2. Modify the extentions.conf file so that it contains the following:<br />

[easycall]<br />

exten => s,1,Answer()<br />

exten => s,2,Dial(...)<br />

3. Run the following <strong>Asterisk</strong> CLI commands:<br />

>sip reload<br />

>dialplan reload


4 EasyCall Gateway <strong>Telephony</strong> <strong>System</strong> Integrator’s <strong>Guide</strong><br />

Integrating with trixbox<br />

If you use the trixbox user interface, use the following procedure to enable the<br />

telephony system to accept connections from the EasyCall Gateway.<br />

To allow the EasyCall Gateway as an incoming peer to trixbox<br />

1. In the trixbox FreePBX interface, click the General Setting link in the left<br />

navigation area.<br />

2. Set Allow Anonymous Inbound SIP Calls to Yes.<br />

The trixbox will allow requests from all SIP servers.<br />

3. Submit the change.<br />

That change updates extensions_additional.conf with the parameter:<br />

ALLOW_SIP_ANON = yes.<br />

Verifying the Integration<br />

Call Detail Records<br />

You are responsible for ensuring that the parameters you set will work in your<br />

environment. While the integration information in this guide is intended as a general<br />

guideline, each environment differs and the information in this guide might not be<br />

accurate or complete for your site.<br />

After completing the integration, install EasyCall on one or more user PCs and place<br />

different types of calls (domestic, international).<br />

A Call Detail Record (CDR) is a file that contains information about calls placed<br />

through the EasyCall Gateway. Information provided in the CDRs can be used for<br />

billing calls to cost centers and for auditing potential abuse.

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