Introduction to Asterisk

Mark Turner

Siteseers Inc.

www.siteseers.com

12 June 2004

Goals

To become familiar with Asterisk's purpose and organization

To step through configuring phone service using Asterisk

Glossary

trunk - a phone line from the Central Office

station - an internal phone

FXO - "Foreign eXchange Office." phone interface to public telephone network (PSTN)

FXS - "Foreign eXchange Station." internal phone interface. "Drives a phone."

SIP - Session Initialization Protocol

IAX - Inter - Asterisk eXchange

Glossary

dialplan - tells a phone system what to do with dialed digits

softphone - a phone implemented in software (i.e., Xten's X-Lite)

hardphone - a SIP-capable telephone (such as the Cisco 7940)

terminal adapter - a device which converts a standard phone into a SIP phone

context - a way of partitioning phone calls. Can be used for security and/or organizational purposes

codec - COder/DECoder. Compresses or expands binary data

Glossary

PSTN - Public Switched Telephone Network - the traditional way of getting dialtone

POTS - Plain Old Telephone System

Phone Switch - a computer which connects calls (the "operator"). Asterisk can be one.

Rate Center - a geographic location used to determine distance-related tolls (quaint?)

Glossary

TDM - Time Division Multiplexing. A way of compressing many calls onto one circuit (T1)

ISDN - Integrated Services Digital Network ("It Still Doesn't Work."). A 2 channel TDM digital line once marketed as the next generation phone service. Achieved blazing speeds of 128Kbps!

PBX - "Private Branch eXchange." A computer which connects calls (see "switch.") Asterisk is a PBX.

Codecs

G.729 - high-quality, low footprint codec. $10 Asterisk implementation. ~8k bps

G.711 - uncompressed channel. Uses ~64k bps

GSM - GSM compression. Decent quality, lower footprint ~13.2k bps

G.723.1 - proprietary. Unsupported in Asterisk other than passthrough. ~6k bps

G.726 – variable bandwidth. Not widely used?

ILBC – open source codec. 15.2k bps

A Telephony Primer

Simple technology

Over 100 years old

Smarts in the switch

New Twists

Multiplexing

Fiber/Digital (goodbye microwave & satellite)

VoIP

Types of VoIP

H.323 – Original. High overhead. Complex. Bloatware?

MGCP – Media Gateway Control Protocol. SIP competitor meant to supercede H.323.

SIP – Session Initialization Protocol. HTTP for the phone. Extensible. NAT-unfriendly.

IAX – Inter-Asterisk eXchange – Asterisk's own. Even lower overhead than SIP. NAT-friendly.

SIP Providers

Free (no PSTN access): Free World Dialup (FWD), Skype, MSN Messenger

Commercial: Time-Warner, Packet8, VoicePulse, Vonage, FeatureTel, IConnectHere

Many, many others.

How To Choose A Provider

Quality (You Get What You Pay For)

Your calling patterns

Supported Rate Centers

Support for your hardware

Price

What Is Asterisk?

Software Phone switch (PBX)

Mark Spencer/Digium

Zaptel hardware

Supported Phones

Most any SIP phone (Cisco 79xx)

Most any SIP ATA (Packet8's DTA-310, Sipura SPA-2000)

Softphones (X-Lite, LIPZ4, Kphone, Windows Messenger)

Asterisk APIs

Channel API

Codec API

File Format API

Application API

File layout

/etc/asterisk

/var/lib/asterisk

/var/log/asterisk

Binaries in /usr/bin & /usr/sbin

Asterisk Dialplan

extensions.conf

format

contexts

Asterisk voicemail

voicemail.conf

file formats

email feature

Practical Examples

voicemail.conf

file formats

email feature

Resources

irc.freenode.net #asterisk

www.asterisk.org

www.digium.com

www.voip-info.org

www.pulver.com/fwd

www.voxilla.org

connect.voicepulse.com

Asterisk's Mailing List

Question and Answer

2004-06-12T00:02:06