Real-time multimedia and communication in
packet networks
Asterisk
The open source IP PBX
Some House Rules
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Practical component of the course
Workings and power of asterisk, an IP Private Branch eXchange (PBX)
Small tutorials will be given on a daily basis before each lecture
Large practical – write your own application that adds value to an
Asterisk PBX
• This will be demonstrated to the class at the end of the course.
• Practical to be done on a Linux machine you can ssh into cc
Some Admin
You should have by now:
- found your extension on pbx.ict.ru.ac.za
- registered on iLanga your two phones (sj and hardphone)
- explored the messaging between the two phones and the SIP proxy
server in iLanga at least in these situations: 1. Registration 2. Call
establishment
with callee answering and without answer (voicemail)
- checked the media stream in iLanga and discovered possible difference
with respect to the case of calling directly an end point.
What happens if you call a telephone registered with iLanga directly via its
IP number?
Some House Rules
•
•
•
•
Practical component of the course
Workings and power of asterisk, an IP Private Branch eXchange (PBX)
Small tutorials will be given on a daily basis before each lecture
Large practical – write your own application that adds value to an
Asterisk PBX
• This will be demonstrated to the class at the end of the course.
• Practical to be done on a Linux machine you can ssh into cc
What is Asterisk (*)?
• A private or enterprise grade exchange generally referred to as a private
branch exchange (PBX)
• Designed to interface telephony hardware or software with any
telephony application seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP, IAX, H.323)
• Private Branch eXchange (PBX)
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?
What is Asterisk (*)?
• A private or enterprise grade exchange generally referred to as a private
branch exchange (PBX)
• Designed to interface telephony hardware or software with any
telephony application seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP, IAX, H.323)
• Private Branch eXchange (PBX)
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?
What is Asterisk (*)?
• A private or enterprise grade exchange generally referred to as a private
branch exchange (PBX)
• Designed to interface telephony hardware or software with any
telephony application seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP, IAX, H.323)
• Private Branch eXchange (PBX)
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?
Asterisk – Supported
Communication Technologies
• Asterisk is designed to allow new interfaces and technologies to be
added easily
• Asterisk’s goal is to support every kind of telephony technology possible
• Asterisk interfaces divided into 3:
• Zaptel hardware
• Non-Zaptel hardware
• Packet voice
Zaptel Hardware
• Check out http://www.zapatatelephony.org/
• Provide integration with traditional and legacy analogue and digital
telephone interfaces
• Zaptel interfaces available from Digium (www.digiumcards.com)
• Zaptel interfaces available for a number of telephony interfaces
• ISDN Basic Rate Interface (BRI)
• ISDN Primary Rate Interface (PRI)
• Analog FXS interface – connect to a station i.e. analogue phone
• Analog FXO interface – connect to an office i.e. PBX
Zaptel Hardware
Digium 4 x FXS card $342 USD
Digium 2 x FXS, 2 x FXO card $360 USD
Digium BRI card $469 USD
Digium BRI card $1345 USD
Packet Voice Protocols
• Standard protocols for communication over packet networks
• Only interfaces that do not require specialised hardware
• E.g.
• SIP
• IAX
• H.323
• IAX
• MGCP
Asterisk’s Architecture
Modules and Applications
• Asterisk’s core contains several engines that play a critical role in the
software’s operation
• At startup, Dynamic module loader loads various modules for:
• Channel drivers
• File formats
• Codecs
• Applications
• Custom applications launcher i.e. the iLanga Prepaid Application
• Asterisk’s switching core accepts calls from any of the various interfaces
and routes them according to the dialplan
• Codec translator permits channels which are compressed with different
codecs to talk to each other
• Scheduler and IO Manager which can be used by applications and
drivers
Asterisk’s Architecture
• Modular API for Asterisk responsible for Asterisk’s
success
• Channel API, File Format API, Codec API,
Application API
Some Asterisk configurations
(basic)
• Asterisk box contains
• 1 analog interface for telephone (FXS interface)
• 1 analog interface to PSTN (FXO interface)
• Ethernet interface for VoIP
Some Asterisk configurations
• Asterisk box contains
• One E1 or (PRI) interface connected to a digital to analog converter
or channel bank
• 15 phones connected channel bank
• 15 lines to PSTN (i.e. Telkom)
Some Asterisk configurations
• In this example we illustrate the possibility of distributing a number of
Asterisk boxes
• Each Asterisk box can be interconnected using
• TDM technology e.g. BRI or PRI
• Data technology/VoIP e.g. Inter Asterisk Exchange (IAX)
Asterisk Filesystem
Organisation
• /etc/asterisk
• Contains Asterisk configuration files – NB directory
• /usr/sbin
• Contains Asterisk binaries
• /usr/lib/asterisk/modules
• Contains runtime modules for channel drivers, codecs, file formats,
applications
• /usr/include/asterisk
• Contains Asterisk C header files for the building the software
• /var/lib/asterisk/agi-bin
• Location of Asterisk Gateway Interface (AGI) for use in dialplan
Asterisk Filesystem
Organisation
• /var/lib/asterisk/astdb
• Asterisk internal database
• Roughly equivalent to Windows registry
• /var/lib/asterisk/mohmp3
• Storage directory mp3s – used for music on hold
• /var/lib/asterisk/sounds
• Storage directory for Asterisk audio files e.g. voice prompts to be
used in IVR menus
• /var/spool/asterisk/outgoing
• Spooling directory for making outgoing calls
• Can be used for callback function
• /var/spool/asterisk/voicemail
• Storage directory for Asterisk voicemail boxes, announcements, etc
Asterisk Channels
• Channel naming convention in Asterisk is standard
• Outgoing channel names (used in Dial application) named in format:
• <technology>/<dialstring>
• <technology> represents type of interface you want to address or use
• E.g. Zap, SIP, IAX2, etc
• <dialstring> is a driver-specific string representing destination desired
Asterisk Channels (Zap)
• <technology>/<dialstring>
• Zap / [g] <identifier>
• <identifier> = number of the channel you are trying to address
• If <identifier> prefixed by ‘g’ then number is interpreted as a group
instead of as a channel
• e.g.
• Zap/g1/0027466223458 (Any available line in group 1)
• Zap/1/0027466223458 (TDM channel 1)
Asterisk Channels (SIP)
• Outgoing channels typical of the form
• SIP / [exten@] <domain> [:<portno>]
• E.g.
• SIP/mos
• SIP:3000@sip.ict.ru.ac.za
• SIP/3000@sip.ict.ru.ac.za:5060
Asterisk Channels (IAX)
• IAX2 / [<user> [:<secret>] @] <domain> [:<portno>]
[/<exten>[@<context>][/<options>]]
• Where <user> and <secret> are optional username and secret to
connect to the host identified by <peer> and
• <portno> = optional port number,
• <exten> = specific extension at an optional context <context>, and
optionally with <options> connection options
• E.g
• IAX2/authname:secretpass@voiptalk.org/442081234567@default
• Call to voiptalk.org using “authname” as username and “secretpass”
as password, and requesting extension 442081234567 in default
context
Running Asterisk and
Environment
• Asterisk can be run in console mode or as a daemon process
• E.g. asterisk –vvvgc (console mode with verbose=3 debugging
• Asterisk (daemon) – started by typing asterisk
• Please always run asterisk as daemon and connect to daemon process
using:
• asterisk –r
• asterisk -vvvvvr
• When connecting to daemon process you will be connected to the
command line interface of Astrerisk (CLI)
• vitalstatistix*CLI>
Asterisk CLI
• When connected to the Asterisk CLI there are a number of commands
you can use
• Go and test them out, see what they do, familiarise yourself with the
environment
• E.g.
• ‘help’
• ‘show applications’
• ‘show application x’
• ‘show codecs'
• ‘show translation'
• ‘extensions reload’
• ‘sip reload’
• CLI include command completion via the tab key
sip.conf
• Please set your phones up to connect to your development box
• Box
• IP = 146.231.121.165
Create a sip.conf file in your home directory, you can use the reference
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
Tomorrow’s Tutorial
1) Create an account for your phone
2) Play around with some of the settings in the sip.conf file