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An Introduction to the Asterisk Open Source PBX. Presented by: Gregory Boehnlein Vice President of N2Net, A New Age Consulting Service, Inc. Company. Hello Class!. Contact Information. Email: [email protected] IRC: Damin on #asterisk

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an introduction to the asterisk open source pbx

An Introduction to theAsterisk Open Source PBX

Presented by:

Gregory Boehnlein

Vice President of

N2Net, A New Age Consulting Service, Inc. Company

contact information

Contact Information

Email: [email protected]

IRC: Damin on #asterisk

Feel free to personally ask me questions or drop me an Email

a little about me

A Little About Me

  • I was born to a family of Orangutans in the Jungles of Borneo
  • Co-Owner of N2Net, a provider of Mission Critical Hosting Services in Cleveland, Ohio celebrating our 10th anniversary
  • Active Open-Source Developer and Activist w/ interest in the Linux and Asterisk development communities
a little about me1

A Little About Me

  • Asterisk Developer and Bug Marshall
  • Primary Maintainer of the AstWind (Asterisk on Windows Project, and no, I’m not kidding)
  • Maintainer of Legacy RedHat Asterisk RPMS
  • I use Asterisk every day, both at Work and at Home


Introduction to VoIP

Introduction to Asterisk

Wrap Up


an introduction to voip

An Introduction to VoIP

What is VoIP?

Voice Over IP

Sending Voice over Internet Protocol

How VoIP works

Continuously sample analog audio (20 ms)

Convert audio into to a digital signaling format or codec

Send digitized stream across the Network as IP packets

Decode the stream to analog for playback


Basic VoIP Terminology

VoIP = Voice Over Internet Protocol

PSTN = Public Switched Telephone Network (AKA Ma Bell, or The Great Satan)

Codec = A Digital Signaling Format

SIP = Session Initiation Protocol

IAX2 = Inter Asterisk Exchange Protocol


VoIP Hardware 101

Proxy = Connects Endpoints Together

Registrar = Authenticates Users

Media Gateway = Translates between the PTSN and Packet Networks

Application Server = Think Webserver

ATA = Analog Telephony Adapter


Why is VoIP Relevant to Consumers?

The Great Myth

“If I switch to VoIP I’ll get Free Long Distance”

Don’t Believe the Hype

The Reality

Trade off of Quality and Reliability for Features

Portability / Flexibility

Cost Effectiveness

More Choice and Control

Every Dollar spent on VoIP goes further


Why is VoIP Relevant to Your Business?

New Revenue Streams

Internet Telephony Service Provider

Managed Voice Applications and Services

Disaster Recovery for Traditional PBX

Hosted PBX Services


Why is VoIP Relevant to Your Business?

Convergence is happening all around you

There are implementation, management and maintenance opportunities for consulting companies.

In 3 years, traditional PBX and Telephone systems will be a thing of the past

Easy Target - Customers are being saturated w/ VoIP Advertising from the likes of Vonage

If you don’t provide the service to them, then someone else will


There Has To Be A Catch

VoIP vs. VoPI

Voice over Public Internet is uncontrollable once it leaves your network

FCC E911 Requirements (As of 5/19/2005)

Must deliver to correct 911 PSAP no matter what, where and how

Ridiculously vague and short sighted

Customer Expectations

Extremely High

The traditional PSTN “just works”

Can’t figure out the “Send” button


There Has To Be A Catch

Competition from the RBOCS and the LECS

Do you really expect Ma Bell to sit idle while ITSPs siphon off their revenue streams?

Virtually unlimited budgets

Lobbying activities in Washington



What Is Asterisk?

This guy, right here!

When: 1999

Who : Mark Spencer

Why : “I needed a phone system and with as small a startup budget as I had for Linux Support Services, I wasn\'t about to buy one, so building one seemed a logical way to go.”


What Is Asterisk?

Officially, Asterisk is an Open Source hybrid TDM and packet voice PBX and IVR platform with ACD functionality. Unofficially, Asterisk is quite possibly the most powerful, flexible, and extensible piece of integrated telecommunications software available.

Its name comes from the asterisk symbol, *, which represents a wildcard, matching any filename.

Similarly, Asterisk the PBX is designed to interface any piece of telephony hardware or software with any telephony application, seamlessly and consistently.


What Is Asterisk?

An Open Source Telephony Swiss Army Knife

A Linux Based PBX w/ Minimal Hardware Reqs

A Community Driven Development Project

A Really, Really Disruptive Technology

Asterisk is any call, any time, from anywhere to anywhere else


Licensing Model

Released and developed under GPL, but Digium retains rights to code-base

All developers submit disclaimers to their code before patches are accepted, allowing for Digium to license specific branches for Commercial projects

This dual-licensing allows companies to purchase license rights to snapshots of the Asterisk codebase to be used in commercial, non-gpl products


Who Is Digium

From Wikipedia, the free encyclopedia.

Digium is the primary developer and sponsor of Asterisk™, The Open Source PBX. Digium offers a variety of specially designed low and high density telephony hardware and professional services related to Asterisk. The company is based in Huntsville, Alabama. Digium sells telephony hardware and provides contract services for operating IP based telephony solutions.


Real World Applications

Key System or PBX Replacement

Voicemail Server

Conferencing Server

Call Center ACD Queue

SIP/H323/MGCP Endpoint for IP Phones

Confound and Confuse Telemarketers

Prank Friends with Random Sound Files

Calling Card Application

Predictive Dialer

Home Answering Machine

which is remarkably similar to
Which is remarkably

similar to……


The Asterisk Development Model

Similar to Linux

Mark Spencer == Linus Torvalds

Core developers with CVS commit rights

1.0 – EOL (Serious Bug Fixes Only)

1.2 – Trunk managed by Drumkilla (Russel Bryant)

Digium employs a handful of full-time developers to just work on the code

Community Supported

Asterisk-dev Mailing list

Weekly Developer Conferences

Held Online Using a “Meet-Me” Bridge

meet the developers

These guys, right here!

Meet the Developers


Under The Hood

Modular architecture like Linux kernel or Apache

Console Interface for debugging / status

Most components can be loaded and unloaded from the CLI

Configuration of system is flexible;

Traditionally using Text Files (/etc/asterisk/ directory)

1.2 provides Real-Time Configuration w/ Database Backend


The Channel API

Channel API Interfaces w/ Hard/Software

Zap – Zaptel Channel Driver

Digium TDM Cards

Zapata Telephony Project Designs

IAX2 – InterAsterisk eXchange Protocol Version 2

Extremely efficient, very simple, voice optimized protocol

Can transport up to 3x as many calls per Megabit than SIP

SIP – Strives to maintain RFC 3261 compatibility

Communicates with SIP Gateways / Phones

Probably the most compatible SIP stack out there despite the overwhelming complexity of the code

H323 – Based on OpenH323

Communicates with H323 Gateways / Phones


The Channel API

Channel API Interfaces w/ Hard/Software

MGCP – Media Gateway Control Protocol

Communicates with MGCP Gateways / Phones

SCCP – Cisco Proprietary Skinny Control Protocol

Communicates with Cisco SCCP Equipment

OSS – Open Sound System

Older Linux Sound Drivers

Communicates with Soundcards

ALSA – Advanced Linux Sound Architecture

New Linux Sound Drivers

Communicated with Soundcards


The Codec Translation API

Codec Translation API Converts Audio Codecs

G.711 Ulaw/Alaw

Ulaw is used in the states, Alaw in Europe

G.726 32Kbps


Requires a license ($10 / channel from Digium)

Most widely deployed, low bandwidth codec (8kbps)



LPC10 (not recommended!)


Open Source, Royalty Free, configurable 4-48kbps, VBR, ABR


The File Format API

This API Allows Reading/Writing of Various File Formats

Some applications may need to archive digital audio streams in different formats

Used by many applications such as VoiceMail, which records messages to disk in whatever format you choose

Available Formats






The Application API

Applications Perform Functions

Modules of code that are used by the Dial Plan

For Example:

Answer: Answer a channel if ringing

BackGround: Play a file while awaiting DTMF tones

Busy: Indicate busy condition (normal busy)

Congestion: Indicate congestion (fast busy)

Dial: Place a call and connect to the current channel

Directory: Provide directory of voicemail extensions

MeetMe: MeetMe conference bridge

MP3Player: Play an MP3 file or stream

MusicOnHold: Play Music On Hold indefinitely

Record: Record to a file

VoiceMail: Leave a voicemail message

VoiceMailMain: Enter voicemail system


The Application API

Simple Dial Plan Example

Dial 216-920-3111 w/ your Cell Phone to hear it live

; CallerID Identify

exten => 2169203111,1,Answer

exten => 2169203111,2,Wait(2)

exten => 2169203111,3,Playback(channel-insecure-warn)

exten => 2169203111,4,SayDigits(${CALLERIDNUM})

exten => 2169203111,5,Wait(1)

exten => 2169203111,6,SayDigits(${CALLERIDNUM})

exten => 2169203111,7,Wait(1)

exten => 2169203111,8,Playback(goodbye)

exten => 2169203111,9,Hangup


Console Output

-- Executing Answer("Zap/1-1", "") in new stack

-- Accepting call from \'2164114184\' to \'2169203111\' on channel 0/1, span 1

-- Executing Wait("Zap/1-1", "2") in new stack

-- Executing Playback("Zap/1-1", "channel-insecure-warn") in new stack

-- Playing \'channel-insecure-warn\' (language \'en\')

-- Executing SayDigits("Zap/1-1", "2164114184") in new stack

-- Executing Wait("Zap/1-1", "1") in new stack

-- Executing SayDigits("Zap/1-1", "2164114184") in new stack

-- Executing Wait("Zap/1-1", "1") in new stack

-- Executing Playback("Zap/1-1", "goodbye") in new stack

-- Playing \'goodbye\' (language \'en\')

-- Executing Hangup("Zap/1-1", "") in new stack

== Spawn extension (inbound, 2169203111, 9) exited non-zero on \'Zap/1-1\'

-- Hungup \'Zap/1-1\'



API Access


Accessible via standard ANSI C

Pre-existing example code for applications, channel drivers etc..

Forms the Core of Asterisk

Well documented, just read the code ;)



Similar to mod_perl for Apache

Single Perl interpreter is loaded and used to process requests

Allows embedding of perl commands directly in Dial Plan

For the more adventurous, can be used to extend Asterisk to unimaginable tasks

Available as part of the asterisk_addons package from CVS

res js similar to res perl except for javascript available from http www pbxfreeware org

Similar to res_perl, except for Javascript

Available from



Asterisk Gateway Interface

Similar to CGI

Write in whatever you want (Perl, PHP, Python, Pascal, Java, BASH… )

Variables are passed on StdIn to your Applications, results and commands are passed back on StdOut

Included w/ Asterisk, no additional work required


Manager API

Allows client/server interaction over TCP/IP sockets w/ authentication

Can be used to issue commands or monitor PBX events

Used by applications such as the Flash Operator Panel and IP Switchboard


Pre-Recorded Prompts

Hundreds of professionally recorded prompts

Recorded by Allison Smith, a Voice Over Professional

Clients include;


Bell Canada



Victoria Secret

Can do custom work hourly or using a credit system

For more information visit:


Connecting Asterisk To The World

Many, Many Options..

TDM Cards from Digium

VoIP Softphones

VoIP Hardware from Various Vendors

VoIP Termination / Origination Service from Carriers

The “ITSP” Internet Telephony Service Provider

sip hardware phones
SIP Hardware Phones

Cisco 7960

Polycom IP-600

iax2 software phones
IAX2 Software Phones




Where To Go For More Information

Digium Website at

Asterisk Website at

Asterisk Docs Project at

VoIP Info Wiki at

Bug Tracker at

#asterisk on

How You Can Help

Get Involved

Try it out

Report Bugs

Make Suggestions

Submit Patches

Help Review, Revise Documentation

contact information1

Contact Information

Email: [email protected]

IRC: Damin on #asterisk

Feel free to personally ask me questions or drop me an Email

(Thanks for Listening!)