1. 1 Kommunikatsiooniteenuste arendus IRT0080 Loeng 10, “telefonisõlmed” Avo Ots telekommunikatsiooni õppetool, TTÜ raadio- ja sidetehnika inst. avo.ots@ttu.ee

2. 2 Voice Telephony & Voice Mail Unified Messaging Conferencing Find-Me-Follow-Me Call Center Video Messaging/Mail Conferencing Surveillance Collaboration Instant Messaging eMeetings Web Conferencing Business Apps CRM, Supply Chain, Call Center Integration WEB-Portal, Desktop, Devices and Server-Business Process Presence

3. 3 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.

4. 4 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

5. 5 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

6. 6 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)

7. 7 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)

8. 8 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})

9. 9 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

10. 10 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.

11. 11 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

12. 12 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

13. 13 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.

14. 14 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

15. 15 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

16. 16 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

17. 17 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

18. 18 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

19. 19 SIP Specifications Supported RFC 3261 - Session Initiation Protocol RFC 3265 - SIP-Specific Event Notification RFC 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-01 draft-burger-sipping-netann-05 draft-ietf-sipping-dialog-package-01 draft-ietf-sipping-service-examples-04

20. 20 Lingid http://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/

21. 21 Presentatsioonid http://ws.edu.isoc.org/data/2006/1267554 9354482287a4f488/telephony.ppt http://www.educause.edu/upload/presenta tions/E06/SESS072/Production%20Qua lity%20Open%20Source%20VoIP.ppt