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Kommunikatsiooniteenuste arendusIRT0080

Loeng 10, “telefonisõlmed”

Avo Otstelekommunikatsiooni õppetool,

TTÜ raadio- ja sidetehnika inst.avo.ots@ttu.ee

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VoiceTelephony & Voice Mail

Unified MessagingConferencing

Find-Me-Follow-MeCall Center

VideoMessaging/MailConferencing Surveillance

CollaborationInstant Messaging

eMeetingsWeb Conferencing

Business AppsCRM, Supply Chain, Call

Center

IntegrationWEB-Portal, Desktop, Devices and

Server-Business Process

Presence

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What is Asterisk…• 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.

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Goals of Asterisk• 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

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Channel types: VoIP• 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

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Channel types - non-VoIP• 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)

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Some Applications• 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)

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Some Applications (cont’d)• Playback(filename)

– Plays a sound file in .gsm format

• Background(filename)– 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})

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Some Applications (cont’d)• MeetMe(conf#)

– Adds the caller to a conference room (optionally muted or unmuted)

• Monitor– Records channel (in and out) to .wav or .gsm

files

• PrivacyManager– Forces anonymous calls to enter valid ANI

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Some Applications (cont’d)• DISA

– Lets callers from one channel get dialtone on another channel

• SetMusicOnHold– You can specify .mp3 files as music on hold

selections (random or sequential)

• MP3Player– Fairly useless, but fun. You can specify files or

streams of .mp3 to be played to callers.

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Some Applications (cont’d)• 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

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Voicemail• 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

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Practical Uses (home)• 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.

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Practical Uses (office)• 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

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Advanced Topics• 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

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Really Advanced Topics• 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

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Really Advanced Topics(cont’d)• 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

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Crazy Extra Stuff That Didn’t Fit• 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

–E.G. 2.1.2.1.2.5.4.3.0.5.1.e164.arpa.

• TDM over ethernet for front-end processing

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SIP Specifications Supported

RFC 3261 - Session Initiation Protocol

RFC 3265 - SIP-Specific Event NotificationRFC 2833 - RTP Payload for DTMF Digits

RFC 2327 - SDP: Session Description Protocol

RFC 3264 - An Offer/Answer Model with SDP

RFC 1889 - RTP: Transport Protocol for Real-Time Applications

RFC 1321 - MD5 Message-Digest Algorithm

RFC 2617 - HTTP Authentication: Basic and Digest Access Authentication

RFC 783 - TFTP Protocol (used for transferal of configuration files to the gateway)

RFC 2705 – Media Gateway Control Protocol (used for Digit Map implementation)

draft-ietf-sipping-mwi-01 - Message Waiting Indication

draft-ietf-sip-refer-07-Refer-To Header

draft-ietf-sipping-cc-transfer-01draft-burger-sipping-netann-05

draft-ietf-sipping-dialog-package-01draft-ietf-sipping-service-examples-04

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Lingidhttp://nerdvittles.com/http://www.trixbox.org/

http://www.counterpath.com/

http://www.loligo.com/asterisk/

http://www.onlamp.com

http://www.voip911.gov/

http://www.e164.org/

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Presentatsioonidhttp://ws.edu.isoc.org/data/

2006/12675549354482287a4f488/telephony.ppt

http://www.educause.edu/upload/presentations/E06/SESS072/Production%20Quality%20Open%20Source%20VoIP.ppt