Asterisk PBX Presentation, 2006

Open Source Software PBX

Хувийн танилцуулга

Erkhembaatar Mungungadas

System Research, System Administrator

Юунд ашиглах вэ?

• Олон улсын телефон яриа

Customer Service / Small Office Home Office

Customer Service / Small Office Home Office

Агуулга

• 1. VoIP технологийн танилцуулга – background, audio codec, protocol

• 2. Asterisk PBX танилцуулга– history, features, platforms

• 3. Asterisk PBX хэрэглээ– Hardware, Telephone Devices, Installation, Configuration

Channels, Phones, Video TransmissionDialing Plans/IVR, AGI, Asterisk-IM, Billing System

VoIP технологийн танилцуулга

• Background: – PBX (Private Branch Exchange)

• Audio codec:– gsm, ulaw, alaw, g729

• Protocol:– SIP, IAX, H323


PBX (Private Branch Exchange)Features- Establishing connections

(circuits) between the telephone sets of two users

- Channeling voice signals between the users

- Providing information for accounting purposes


• PBX (Private Branch Exchange)


Audio codec(COmpression/DECompression)


Protocols

• SIP (Session Intitation Protocol)

• IAX (Inter-Asterisk Exchange)

• H.323 (ITU standard)


• SIP (Session Intitation Protocol)– SIP, the session intitation protocol, is the IETF protocol for VOIP and

other text and multimedia sessions, like instant messaging, video and other services.


IAX (Inter-Asterisk Exchange)

developed by Digium for the purpose of communicating with other Asterisk servers (hence “the Inter-Asterisk eXchange protocol”).

IAX is a transport protocol that uses a single UDP port (4569) for both the channel signaling and RTP streams.


H.323 IP transport mechanism for video-conferencing.

Asterisk PBX танилцуулга

• History

• Features

• Platform


History

Creator Mark Spencer of Digium, Inc.

CentOS based All-in-One

Embedded Asterisk System


Features• Alarm Receiver • Append Message • Calling Cards • Conference Bridging • Distinctive Ring • Protocol Conversion (IAX->SIP)• Roaming Extensions • Supervised Transfer• Transcoding (ulaw->gsm etc.)• Three-way Calling• Voicemail to email (attach audio file)• Web Voicemail Interface • ...


Platform:• Unix / Linux / Mac OS X• Win32 http://www.asteriskwin32.com/

Hardware supported with two channels:

TAPI channel:

support TAPI compatible hardware full duplex & half duplex devices (most of voice modem devices).

CAPI channel:

ISDN device CAPI compatible

http://www.asteriskwin32.com/

Asterisk PBX хэрэглээ

• Hardware Requirement• Telephone Devices• Installation, Configuration• Channels• Phones• Video Transmission• Dialing Plans/IVR, AGI, Festival, Asterisk-IM• Billing System, PC-2-PC, International Calling


Hardware Requirement


Telephone Devices

• Digital Interface Cards • Digium Wildcard TE205P• PCI 5 / E1 and T1/J1 channel bank

• Analog Interface Cards • Digium Wildcard TDM400P• PCI 2.2 / FXS and FXO


Installation• Debian GNU/Linux Sarge ( apt-get install asterisk )

libpri* zaptel apt-get install asterisk

asterisk-chan-capi asterisk-config asterisk-dev asterisk-doc asterisk-h323 asterisk-oh323 asterisk-sounds-main asterisk-web-vmail


Configuration (/etc/asterisk)

• asterisk.conf• modules.conf• extensions.conf• sip.conf• iax.conf• voicemail.conf• manager.conf


ChannelsSIP (sip.conf, extensions.conf)[sipuser]

type=friend

callerid=("SIP User" <1000>)

username=sipuser

exten => s,1,Dial(SIP/{$EXTEN})

IAX / IAX2 (iax.conf, extensions.conf)IAX/[<user>[:<secret>]@]<peer>[:<portno>][/<exten>[@<context>][/<options>]]

IAX/mark:asdf@myserver/6275@default – Call to "myserver" using "mark" as username and "asdf" as password, and requesting extension 6275 in default context


Phones

• Analog phones

• IP phones

• Software phones

X-Lite, DIAX, Kapanga


Video Transmission• Supported Channels :

– SIP, IAX2

• Supported Video Codecs :– H.261, H.263, H.263p

• Supported Video Codecs :– AuPix SIP and H.323 video phone– iFon: for PocketPC PDAs– Microsoft Windows Messenger for Windows PC's– X-Lite, eyeBeam for Windows and OSX– Kapanga Sip Video Phone


Dialing Plansextensions.conf (/etc/asterisk)[general]static=yes writeprotect=yes videosupport=yes

[globals]PHONE1=Zap/1PHONE2=SIP/6002

Extensions[local]

exten => 6601,1,Macro(oneline,${PHONE1}) exten => 6602,1,Macro(oneline,${PHONE2})


IVR (Interactive Voice Responder)• [default]• exten => s,1,Answer• exten => s,2,SetMusicOnHold(default)• exten => s,3,DigitTimeout,5• exten => s,4,ResponseTimeout,10• ;SAI menu - 1 for tech support, 2 for voicemail• exten => s,5,Background(sai-welcome)

• ; Tech Support• exten => 1,1,AGI(dima-test.agi)• exten => 1,2,Queue(rep-tech)

• ; Leave Voicemail• exten => 2,1,VoicemailMain()• exten => 2,2,Hangup


AGI (Asterisk Gateway Interface)The Asterisk Gateway Interface provides external programs may control the Asterisk dialplan. Communicate with relational databases (such as PostgreSQL or MySQL)

/var/lib/asterisk/agibin

Example:• exten => 123,1,Answer( )• exten => 123,2,AGI(agi-test.agi)


• Festival (Text-to-Speech)Festival is a popular open source text-to-speech engine.

• Example: (extensions.conf)• exten => 123,1,Answer( )• exten => 123,2,Festival(Asterisk and Festival are working together)


• Asterisk-IM (Jive Software Integration)• Integration component to Asterisk for Jive's

Jabber/XMPP server


Billing System: Asterisk2Billing• Authenticate with the use of a Cardnumber • Reporting • Rate-engine • Simultaneous access for same card • SIP/IAX Friends Management • post-pay & prepay • Recurring service over the card • Reload Asterisk through UI & manager • IVR Customization - (say balance, forwarding, …)


Test Calling

• AsteriskWin32 PBX

• X-Lite softphone

• Configuration

• VoipJet.com Registration

• Test Calling …

Thank you!

Any questions!

Technology

Asterisk PBX Presentation, 2006