Transcript Asterisk in Three Beer’s Time Or Less

Asterisk in Three
Beer’s Time Or
Less
Or: How I Stopped Worrying and Learned
to Love The Dialtone
© John Todd 2003-08-20 [email protected]
What is Asterisk? (generally)
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Asterisk is a PBX replacement system, designed to
reproduce the features of standard office phone
systems. Asterisk is also a Voice over IP toolkit
which allows interaction between these PBX
features and IP-based networks (local and remote.)
Asterisk is hardware independent, and is designed
to run on OSS operating systems.
© 2003 John Todd ([email protected])
What is Asterisk? (details)
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Telephony gateway (TDM channels - PRI,POTS)
VoIP Gateway (IP channels)
IVR system (Interactive Voice Response)
Voicemail system
Scriptable telephony-to-anything (Perl, C, etc.)
More than will fit on this slide
© 2003 John Todd ([email protected])
What Asterisk isn’t
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A billing system
A CRM system
A web server or XML server (re: Cisco 79xx)
A configuration tool for VoIP devices
A voice recognition system
A USENET or email client
© 2003 John Todd ([email protected])
Goals of Asterisk
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Provide Open-Source implementations of basic PBX
functionality
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
Sell TDM hardware cards for Digium
© 2003 John Todd ([email protected])
Who is Digium?
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Primary supporter of Asterisk development
Owner of the CVS server/bug system/mailing list
boxes/etc.
Approves all patches and features by community
Produces TDM cards (“Wildcard” hardware)
which works particularly well with Asterisk
Owner of the disclaimers for all contributions to
Asterisk, holder of copyright
© 2003 John Todd ([email protected])
WHAT?!? NON-GPL!?? JIHAD!!
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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 contributors saying
that Digium can do so
Generally everyone is happy with this system
Hold your flames until after I’m finished. :-)
© 2003 John Todd ([email protected])
Channel types: VoIP
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SIP - Session Initiation Protocol
H.323
MGCP - Media Gateway Control Protocol
SCCP - Skinny Client Control Protocol (Cisco)
All of these use UDP for setup/transport except for
SCCP, which uses a mix of UDP/TCP
© 2003 John Todd ([email protected])
Channel Types: VoIP (cont’d)
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Phones for VoIP (SIP):
Grandstream 102 - ~$85 new
 Cisco ATA 186 - 2 lines of analog - ~$140 new
 Cisco 7960/7940 - very nice deskphone - ~$300 used
 Many others - market is starting to flood with new
vendors from SE Asia
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Software for VoIP (SIP)
www.xten.com - free SIP client (“Lite”)
 gnophone.com - Linux SIP client
 Windows Messenger - don’t ask me, I don’t know how.
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© 2003 John Todd ([email protected])
Channel types - non-VoIP
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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 (yuck)
Speaker/headphones (don’t try this at home, kids)
© 2003 John Todd ([email protected])
System Requirements
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No clear rule of thumb on processor size; at least 400mhz PIII
recommended
Almost any version of Linux supported; RH 7.x or 8 is dev
platform (9 has tweak issues)
Source + binaries (including sounds) are ~35m
Using complex codecs (i.e.: G.729, speex, etc.) will increase
processor load dramatically
Best to have a >1.5ghz machine for multi-channel use
Linux preferred, though *BSD slowly starting to become
stable for non-hardware channels
© 2003 John Todd ([email protected])
Gotchas
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Mpg123 needs to actually be mpg123, and not
mpg321 (and located in /usr/bin/)
You need to have matching kernel source
installed correctly to compile Asterisk/Zaptel
VoIP isn’t simple the first time you do it
Asterisk documentation is less than adequate mailing list and Google have better data
© 2003 John Todd ([email protected])
Call Flow (briefly)
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Calls come in on channels and are then handed to the
“extensions.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 context name and a dialed number
The dialplan then matches (with modified regexp’s) the
number being dialed, and runs applications accordingly
Each match on the dialed number has an order of steps
called ‘Priorities’, and are indicated with an integral
incrementing number (argh! Like a horrible BASIC
Frankenstein, without the flexibility! )
© 2003 John Todd ([email protected])
Regular expressions (briefly)
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All regular expressions start with “_” character in dial
examinations.
“X” means any number, “N” is any number other than 0
or 1
“.” means any number of characters
Brackets represent groups, with standard “-” and “,”
meanings ([1-9] or [0,1,2])
Better regexp in the works
Example: _1410985012X is the same as
_1410985012[0-9]
© 2003 John Todd ([email protected])
Call Flow (cont’d)
[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
© 2003 John Todd ([email protected])
Redirection based on ANI
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You can match against calling number instead
of called number.
This is a.k.a. “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
© 2003 John Todd ([email protected])
Redirection of Call Flow
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GotoIf - can parse basic Booleans
GotoIfTime - handy way to deal with time-based
redirection
Some applications will add 101 to the existing
priority when certain errors occur (notably, Dial
does this on busy, and DBget/DBput do this on
errors reading from the internal database)
Any other type of errors result in channel
hangup
© 2003 John Todd ([email protected])
Variables
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${VARNAME} is how variables are used
Variables must be declared before Booleans can
be performed (gah - no null value comparitor)
Variables can be nested during setting
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Exten => 123,1,SetVar(BAR=blah)
Exten => 123,2,SetVar(FOO=3)
Exten => 123,3,SetVar(NEWVAR.${FOO} = ${BAR})
This results in ${NEWVAR.3} being set to “blah”
© 2003 John Todd ([email protected])
Special Variables
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${EXTEN} - always the most important variable.
This is the number that is being currently
evaluated.
${CALLERIDNUM} - the ANI (if available) of the
call leg that is creating the call
Some others, less used: ${EPOCH},
${ENV(var)}, ${CONTEXT}, ${PRIORITY},
several other descriptors of the call leg we’re
processing
© 2003 John Todd ([email protected])
Some Applications
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Dial - connects an inbound call with some other
channel. One specifies the technology (SIP,
Zap, H323, etc.) the number to be dialed, the
Ring-No-Answer delay, and options (if desired)
exten => 1234,1,Dial(SIP/1234,25)
exten => 1234,2,Voicemail2(u1234)
© 2003 John Todd ([email protected])
Some Applications (cont’d)
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Playback(filename)
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Plays a sound file in .gsm format
Background(filename)
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Plays a sound file while listening for DTMF (touch
tone) input
[test]
exten => 123,1,Background(press-a-number)
exten => 123,2,Goto(1)
exten => _X,1,SayDigits(${EXTEN})
© 2003 John Todd ([email protected])
Some Applications (cont’d)
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MeetMe(conf#)
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Monitor
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Adds the caller to a conference room (optionally
muted or unmuted)
Records channel (in and out) to .wav or .gsm files
PrivacyManager
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Forces anonymous calls to enter valid ANI
© 2003 John Todd ([email protected])
Some Applications (cont’d)
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DISA
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SetMusicOnHold
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Lets callers from one channel get dialtone on
another channel
You can specify .mp3 files as music on hold
selections (random or sequential)
MP3Player
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Fairly useless, but fun. You can specify files or
streams of .mp3 to be played to callers.
© 2003 John Todd ([email protected])
Some Applications (cont’d)
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There are over 80 different applications in the
system - no time to talk about them all
Applications are easily created and added if
you’re a decent C coder
Channels are generic, so you don’t have to
know about any of the ugly VoIP or TDM stuff
© 2003 John Todd ([email protected])
Voicemail
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Voicemail can be sent as email as well as stored
on disk (1 minute = 100kb)
Short pages can be sent to email addresses
when VM received
Customizable timezones and time readouts per
user - supports multiple languages
.wav, .gsm file storage or email
Dial by name directory hinges on VM data
© 2003 John Todd ([email protected])
Practical Uses (home)
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Ditch your long distance company! SIP long
distance (domestic and int) providers starting to
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 forward callers
who are on “priority” lists based on ANI.
© 2003 John Todd ([email protected])
Practical Uses (office)
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Ditch your LD company (see prior slide)
Interconnect office PBXs at zero network cost
Get “Unified Messaging”
Give ubiquitous access to the PBX for home/travelling
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
© 2003 John Todd ([email protected])
Advanced Topics
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Call queues - you can build a call center with
Asterisk, with various 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 - first answer gets it)
Multi-language support with same dialplan
Festival integration for voice synthesis
© 2003 John Todd ([email protected])
Really Advanced Topics
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Manager interface: TCP socket based interface
for controlling and monitoring the system; meant
for automated manager tools (see: gastman)
AGI scripts: built-in scriptable hooks to allow
passing of data back and forth between Asterisk
and external programs.
Asterisk.pm - Perl module that works with AGI to
handle gruntwork of call handling
© 2003 John Todd ([email protected])
Really Advanced Topics(cont’d)
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Sybase and MySQL modules
CDR (call detail record) output can be
customized or put into database instead of flat
file
Use IAX2 trunk mode to get up to 200% more
calls in the same bandwidth as other VoIP
systems
Route your calls to least-cost providers
© 2003 John Todd ([email protected])
Crazy Extra Stuff That Didn’t Fit
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Can run PPP or HDLC over channels - Asterisk can
be a RAS server or a router (masochism)
Can use speaker/microphone as a “phone line”
Can do video calls or conferencing
ENUM e.164 DNS-based call routing
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2.1.2.1.2.5.4.3.0.5.1.e164.arpa.
TDM over ethernet for front-end processing
© 2003 John Todd ([email protected])
Resources and Wrap-Up
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http://www.asterisk.org/
http://www.digium.com/
http://www.loligo.com/asterisk/
http://www.wwworks-inc.com/asterisk/
http://www.xten.com/
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
John Todd - [email protected]
Asterisk Gun For Hire
PLUG Advanced Topics 2003-08-20
© 2003 John Todd ([email protected])