Real-time multimedia and communication in packet networks Asterisk The open source IP PBX

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

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 (voic ) - 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? Please make sure that problems that you had yesterday are cleared in the first part of the lecture. Enjoy Asterisk!

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 Provide integration with traditional and legacy analogue and digital telephone interfaces Zaptel interfaces available from Digium ( 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/voic Storage directory for Asterisk voic boxes, announcements, etc

Asterisk Channels Channel naming convention in Asterisk is standard Outgoing channel names (used in Dial application) named in format: / represents type of interface you want to address or use E.g. Zap, SIP, IAX2, etc is a driver-specific string representing destination desired

Asterisk Channels (Zap)‏ / Zap / [g] = number of the channel you are trying to address If prefixed by ‘g’ then number is interpreted as a group instead of as a channel e.g. Zap/g1/ (Any available line in group 1)‏ Zap/1/ (TDM channel 1)‏

Asterisk Channels (SIP)‏ Outgoing channels typical of the form SIP / [: ] E.g. SIP/mos

Asterisk Channels (IAX)‏ IAX2 / [ [: [: ] [/ ][/ ]] Where and are optional username and secret to connect to the host identified by and = optional port number, = specific extension at an optional context, and optionally with connection options E.g Call to voiptalk.org using “authname” as username and “secretpass” as password, and requesting extension 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 =  Create a sip.conf file in your home directory, you can use the reference

Tomorrow’s Tutorial 1)Create an account for your phone 2)Play around with some of the settings in the sip.conf file