02 - Asterisk - The Future of Telecommunications (1)

7/31/2019 02 - Asterisk - The Future of Telecommunications (1)

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www.convergencetechnologycenter.orgDUE 402356

AsteriskThe Future of Telecommunications

Vincente DIngianniDirector of Professional Services

Binary Systems, Inc.

[email protected]


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2www.convergencetechnologycenter.org

What is Asterisk?

Asterisk is a complete VoIP Softswitch, designed toreproduce the features of standard office PBX system.

Asterisk is also a Voice over IP toolkit which allowsinteraction between these PBX features and IP-basednetworks (local and remote.)

Asterisk is hardware independent, and is designed to runon numerous operating systems.


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Proprietary API

Media Gateways / Endpoints

MGCP

PCI Bus Ethernet

SCCPH.323

Asterisk

IAX

EthernetEthernet

SIP

Asterisk Softswitch System Architecture


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Asterisk Capabilities

Telephony gateway (TDM channels - PRI,POTS)

VoIP Gateway (IP channels)

IVR system (Interactive Voice Response)

Voicemail System

Scriptable telephony-to-anything (Perl, C, etc.)

much, much more


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Asterisk is not

A Billing system A CRM system A web server or XML server A configuration tool for VoIP devices A voice recognition system A USENET or email client

but it is often bundled with these subsystems to form a complete solution.


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Asterisk Goals

Provide Open-Source implementations of basic PBXfunctionality

Be vendor neutral (despite last point here) Be as all-encompassing as possible for features Be flexible and provide hooks for advanced features Move proprietary hardware features into open source

software (HMP functionality)

Integrate with 3rd

party telephony hardware devices(DSP functionality)

Sell TDM hardware cards for Digium


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Who is Digium?

Primary supporter of Asterisk development. Owner of the CVS server/bug system/mailing

list boxes/etc.

Approves all patches and features bycommunity Produces TDM cards (Wildcard hardware)

which works particularly well with Asterisk

Owner of the disclaimers for all contributionsto Asterisk, holder of copyright


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Asterisk is not quite GPL

Asterisk is GPL, but with an important clause Digium can license branches of the source such

that those branches are not GPL

Digium gets disclaimers from all contributorssaying that Digium can license branches of thecode.


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VoIP Channels

SIP - Session Initiation Protocol (internal stack) H.323 via OpenH323 Project MGCP - Media Gateway Control Protocol

(internal stack) SCCP Cisco Skinny Protocol (internal stack)

IAX Inter-Asterisk eXcange Protocol Special open-source protocol developed for

communicaiton between Asterisk servers.


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VoIP Channel Endpoints

Phones for VoIP (SIP): Grandstream 102 Cisco ATA 186 Sipura Cisco 7960/7940

Polycom IP-501, IP-601, etc. Snome Many others

Software for VoIP (SIP) www.xten.com - free SIP client (Lite) gnophone.com - Linux SIP client Windows Messenger


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TDM and Other Channels

TDM POTS cards (Digium, Zapata, Voicetronix, etc.) TDM Digital (AdTran VoFR, Digium E1/T1, etc.)

All TDM cards require install of Zaptel driver suite

CAPI (ISDN card support for Linux ISDN driver) USB dongle for FXS Modem drivers for certain modems

Speaker/headphones


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System Requirements

No clear rule of thumb on processor size; at least 500 MHzPIII recommended. Almost any version of Linux is supported. Source & binaries (including sounds) are ~35 MB

Using complex codecs (i.e.: G.729, Speex, etc.) will increaseprocessor load dramatically Remember this is processed on the host processor HMP

Best to have a > 1.5 GHz machine for multi-channel use.

Mac OS X / FreeBSD is becoming stable for non-hardwarechannels.

VMWare and Parallels Virtual Machines


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Call Flow (briefly)

Calls come in on channels and are then handed to theextensions.conf file, which is the dialplan

Dialplan contains logical sections of matches called Contexts,

and each channel sends a call into the dialplan with a contextname and a dialed number.

The dialplan then matches (with modified regexps) the numberbeing dialed, and runs applications accordingly

Each match on the dialed number has an order of steps calledPriorities, and are indicated with an integral incrementingnumber.


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Regular Expressions (briefly)

All regular expressions start with _ character in dialexaminations.

X means any number, N is any number other than0 or 1

. means any number of characters Brackets represent groups, with standard - and ,

meanings ([1-9] or [0,1,2]) Example: _1410985012X is the same as

_1410985012[0-9]


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Call Flow (contd)

[from-my-pri]

exten => 14109850123,1,Answer

exten => 14109850123,2,Wait(2)

exten => 14109850123,3,Playback(monkeys)exten => 14109850123,4,Goto(more-monkeys,123,1)

[more-monkeys]

exten => _12X,1,Playback(sorry-no-more-monkeys)exten => _12X,2,Hangup


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Redirection based on ANI

You can match against calling numberinstead of called number.

This is known as The ex-girlfriend filter by

the inventor of the routines This pattern matches against called number

(1410) and also against calling numer (301)

exten => 14109850123/3013659999,1,Busy


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Redirection of Call Flow

GotoIf - can parse basic Booleans GotoIfTime - handy way to deal with time-based

redirection

Some applications will add 101 to the existingpriority when certain errors occur (notably, Dialdoes this on busy, and DBget/DBput do this onerrors reading from the internal database)

Any other type of errors result in channelhangup


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Variables

${VARNAME} is how variables are used Variables must be declared before Booleans

can be performed

Variables can be nested during settingexten => 123,1,SetVar(BAR=blah)

exten => 123,2,SetVar(FOO=3)

exten => 123,3,SetVar(NEWVAR.${FOO} = ${BAR})

This results in ${NEWVAR.3} being set to

blah


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Special Variables

${EXTEN} - always the most importantvariable. This is the number that is beingcurrently evaluated.

${CALLERIDNUM} - the ANI (if available) ofthe call leg that is creating the call Some others, less used: ${EPOCH}, $

{ENV(var)}, ${CONTEXT}, ${PRIORITY},

several other descriptors of the call leg wereprocessing


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Some Applications

Dial - connects an inbound call with someother channel.

The first argument specifies the technology

(SIP, Zap, H323, etc.) and the number to bedialed, the Ring-No-Answer delay, andoptions (if desired)

exten => 1234,1,Dial(SIP/1234,25)exten => 1234,2,Voicemail2(u1234)


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Some Applications (contd)

Playback(filename) Plays a sound file in .gsm format

Background(filename) Plays a sound file while listening for DTMF (touch

tone) input

[test]

exten => 123,1,Background(press-a-number)

exten => 123,2,Goto(1)exten => _X,1,SayDigits(${EXTEN})


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MeetMe(conf#) Adds the caller to a conference room (optionally

muted or unmuted)

Monitor Records channel (in and out) to .wav or .gsm files

PrivacyManager

Forces anonymous calls to enter valid ANI


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DISA Lets callers from one channel get dialtone on

another channel

SetMusicOnHold You can specify .mp3 files as music on hold

selections (random or sequential)

MP3Player Fairly useless, but fun. You can specify files or

streams of .mp3 to be played to callers.


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There are over 80 different applications in thesystem more created each day.

Applications are easily created and added if

youre a decent C coder or scripting coder. Channels are generic, so you dont have to

know about any of the ugly VoIP or TDM

stuff.


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Voicemail

Voicemail can be sent as email as well asstored on disk (1 minute = 100KB)

Short pages can be sent to email addresseswhen VM received

Customizable timezones and time readouts peruser - supports multiple languages

WAV or GSM file format for storage oremail Dial by name directory hinges on VM data


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Practical Uses

Ditch your long distance company! SIP longdistance (domestic and int) providers startingto crop up at low rates. Use Asterisk to

gateway to them. Prevent phone spam! Callers with no CID get

ditched.

Filter your phone lines. Allow or forwardcallers who are on priority lists based onANI.


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Practical Uses

Enterprise-quality SIP connection services are now available.

Interconnect office PBXs at zero network cost Get Unified Messaging Give ubiquitous access to the PBX for home/traveling employees Disaster recovery scenarios Move phones into your IT department and away from your

expensive PBX consulting firm Eliminate adds/moves/changes as physical chores


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Advanced Topics

Call queues - you can build a call center with Asterisk, withvarious call weightings and agent logins/hot seating

Multi-ring, cascading ring with different technologies (inbound

calls forward to your desk line and your cell phone - firstanswer gets it)

Multi-language support with same dialplan

Festival integration for voice synthesis


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Really Advanced Topics

Manager interface: TCP socket based interface forcontrolling and monitoring the system; meant forautomated manager tools (see: gastman)

AGI scripts: built-in scriptable hooks to allow passingof data back and forth between Asterisk and externalprograms.

Asterisk.pm - Perl module that works with AGI tohandle grunt work of call handling


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Really Advanced Topics (cont)

Sybase and MySQL modules

CDR (call detail record) output can be customized orput into database instead of flat file

Use IAX2 trunk mode to get up to 200% more calls inthe same bandwidth as other VoIP systems

Dynamically Route your calls to least-cost providers


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Other Asterisk Applications

Can run PPP or HDLC over channels Asterisk can be a RAS server or a router

Can use speaker/microphone as a phone line

Video Calls or Conferencing

ENUM e.164 DNS-based call routing

Example: 2.1.2.1.2.5.4.3.0.5.1.e164.arpa.

TDM over ethernet for front-end processing


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Asterisk Resources

http://www.asterisk.org/ - Latest Source Code

http://www.digium.com/ - Asterisk TDM hardware

http://www.voip-info.org/ - General VoIP How-To Info

http://www.xten.com/ - Softphone

http://www.asterisk-vonage.com/ - Asterisk to Vonage connectivity

http://www.binary-systems.com/ - Asterisk Consulting & Training Services
http://www.asterisk.org/http://www.digium.com/http://www.voip-info.org/http://www.xten.com/http://www.asterisk-vonage.com/http://www.binary-systems.com/http://www.binary-systems.com/http://www.asterisk-vonage.com/http://www.xten.com/http://www.voip-info.org/http://www.digium.com/http://www.asterisk.org/

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02 - Asterisk - The Future of Telecommunications (1)