Real-time multimedia and communication in packet networks Asterisk

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 e. Xchange (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 i. Langa your two phones (sj and hardphone) - explored the messaging between the two phones and the SIP proxy server in i. Langa at least in these situations: 1. Registration 2. Call establishment with callee answering and without answer (voicemail) - checked the media stream in i. Langa and discovered possible difference with respect to the case of calling directly an end point. What happens if you call a telephone registered with i. Langa directly via its IP number?

Some House Rules • • Practical component of the course Workings and power of asterisk, an IP Private Branch e. Xchange (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 e. Xchange (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. digium. 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 i. Langa 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 Vo. IP

Some Asterisk configurations • Asterisk box contains • One E 1 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/Vo. IP 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/mohmp 3 • Storage directory mp 3 s – 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: • / • represents type of interface you want to address or use • E. g. Zap, SIP, IAX 2, etc • is a driver-specific string representing destination desired

Asterisk Channels (Zap) • <technology>/<dialstring> • Zap / [g] <identifier> • <identifier> = number 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/g 1/0027466223458 (Any available line in group 1) • Zap/1/0027466223458 (TDM channel 1)

Asterisk Channels (SIP) • Outgoing channels typical of the form • SIP / [exten@] [: ] • E. g. • SIP/mos • SIP: 3000@sip. ict. ru. ac. za • SIP/3000@sip. ict. ru. ac. za: 5060

Asterisk Channels (IAX) • IAX 2 / [ [: ] @] [: ] [/[@][/]] • 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 • IAX 2/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 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