An Introduction to the Asterisk Open Source PBX

1. An Introduction to theAsterisk Open Source PBX Presented by: Gregory Boehnlein Vice President of N2Net, A New Age Consulting Service, Inc. Company

2. Hello Class!

3. Contact Information Email: damin@nacs.net IRC: Damin on irc.freenode.org #asterisk Feel free to personally ask me questions or drop me an Email

4. A Little About Me • I was born to a family of Orangutans in the Jungles of Borneo • Co-Owner of N2Net, a provider of Mission Critical Hosting Services in Cleveland, Ohio celebrating our 10th anniversary • Active Open-Source Developer and Activist w/ interest in the Linux and Asterisk development communities

5. A Little About Me • Asterisk Developer and Bug Marshall • Primary Maintainer of the AstWind (Asterisk on Windows Project, and no, I’m not kidding) • Maintainer of Legacy RedHat Asterisk RPMS • I use Asterisk every day, both at Work and at Home

6. Scary Resemblance, Isn’t It?

7. Summary Introduction to VoIP Introduction to Asterisk Wrap Up Questions

8. An Introduction to VoIP What is VoIP? Voice Over IP Sending Voice over Internet Protocol How VoIP works Continuously sample analog audio (20 ms) Convert audio into to a digital signaling format or codec Send digitized stream across the Network as IP packets Decode the stream to analog for playback

9. Basic VoIP Terminology VoIP = Voice Over Internet Protocol PSTN = Public Switched Telephone Network (AKA Ma Bell, or The Great Satan) Codec = A Digital Signaling Format SIP = Session Initiation Protocol IAX2 = Inter Asterisk Exchange Protocol

10. VoIP Hardware 101 Proxy = Connects Endpoints Together Registrar = Authenticates Users Media Gateway = Translates between the PTSN and Packet Networks Application Server = Think Webserver ATA = Analog Telephony Adapter

11. Why is VoIP Relevant to Consumers? The Great Myth “If I switch to VoIP I’ll get Free Long Distance” Don’t Believe the Hype The Reality Trade off of Quality and Reliability for Features Portability / Flexibility Cost Effectiveness More Choice and Control Every Dollar spent on VoIP goes further

12. Why is VoIP Relevant to Your Business? New Revenue Streams Internet Telephony Service Provider Managed Voice Applications and Services Disaster Recovery for Traditional PBX Hosted PBX Services

13. Why is VoIP Relevant to Your Business? Convergence is happening all around you There are implementation, management and maintenance opportunities for consulting companies. In 3 years, traditional PBX and Telephone systems will be a thing of the past Easy Target - Customers are being saturated w/ VoIP Advertising from the likes of Vonage If you don’t provide the service to them, then someone else will

14. There Has To Be A Catch VoIP vs. VoPI Voice over Public Internet is uncontrollable once it leaves your network FCC E911 Requirements (As of 5/19/2005) Must deliver to correct 911 PSAP no matter what, where and how Ridiculously vague and short sighted Customer Expectations Extremely High The traditional PSTN “just works” Can’t figure out the “Send” button

15. There Has To Be A Catch Competition from the RBOCS and the LECS Do you really expect Ma Bell to sit idle while ITSPs siphon off their revenue streams? Virtually unlimited budgets Lobbying activities in Washington YOU DO NOT OWN THE LAST MILE!!!

16. An Introduction to Asterisk

17. What Is Asterisk? This guy, right here! When: 1999 Who : Mark Spencer Why : “I needed a phone system and with as small a startup budget as I had for Linux Support Services, I wasn't about to buy one, so building one seemed a logical way to go.”

18. What Is Asterisk? Officially, Asterisk is an Open Source hybrid TDM and packet voice PBX and IVR platform with ACD functionality. Unofficially, Asterisk is quite possibly the most powerful, flexible, and extensible piece of integrated telecommunications software available. Its name comes from the asterisk symbol, *, which represents a wildcard, matching any filename. Similarly, Asterisk the PBX is designed to interface any piece of telephony hardware or software with any telephony application, seamlessly and consistently.

19. What Is Asterisk? An Open Source Telephony Swiss Army Knife A Linux Based PBX w/ Minimal Hardware Reqs A Community Driven Development Project A Really, Really Disruptive Technology Asterisk is any call, any time, from anywhere to anywhere else

20. Licensing Model Released and developed under GPL, but Digium retains rights to code-base All developers submit disclaimers to their code before patches are accepted, allowing for Digium to license specific branches for Commercial projects This dual-licensing allows companies to purchase license rights to snapshots of the Asterisk codebase to be used in commercial, non-gpl products

21. Who Is Digium From Wikipedia, the free encyclopedia. Digium is the primary developer and sponsor of Asterisk™, The Open Source PBX. Digium offers a variety of specially designed low and high density telephony hardware and professional services related to Asterisk. The company is based in Huntsville, Alabama. Digium sells telephony hardware and provides contract services for operating IP based telephony solutions.

22. Real World Applications Key System or PBX Replacement Voicemail Server Conferencing Server Call Center ACD Queue SIP/H323/MGCP Endpoint for IP Phones Confound and Confuse Telemarketers Prank Friends with Random Sound Files Calling Card Application Predictive Dialer Home Answering Machine

23. The Asterisk Development Model

24. The Asterisk Development Model

25. Which is remarkably similar to……

26. The Linux Development Model

27. The Asterisk Development Model Similar to Linux Mark Spencer == Linus Torvalds Core developers with CVS commit rights 1.0 – EOL (Serious Bug Fixes Only) 1.2 – Trunk managed by Drumkilla (Russel Bryant) Digium employs a handful of full-time developers to just work on the code Community Supported Asterisk-dev Mailing list Weekly Developer Conferences Held Online Using a “Meet-Me” Bridge

28. High Level Overview of a Developer’s Conference

29. These guys, right here! Meet the Developers

30. Under The Hood

31. Under The Hood Modular architecture like Linux kernel or Apache Console Interface for debugging / status Most components can be loaded and unloaded from the CLI Configuration of system is flexible; Traditionally using Text Files (/etc/asterisk/ directory) 1.2 provides Real-Time Configuration w/ Database Backend

32. The Channel API Channel API Interfaces w/ Hard/Software Zap – Zaptel Channel Driver Digium TDM Cards Zapata Telephony Project Designs IAX2 – InterAsterisk eXchange Protocol Version 2 Extremely efficient, very simple, voice optimized protocol Can transport up to 3x as many calls per Megabit than SIP SIP – Strives to maintain RFC 3261 compatibility Communicates with SIP Gateways / Phones Probably the most compatible SIP stack out there despite the overwhelming complexity of the code H323 – Based on OpenH323 Communicates with H323 Gateways / Phones

33. The Channel API Channel API Interfaces w/ Hard/Software MGCP – Media Gateway Control Protocol Communicates with MGCP Gateways / Phones SCCP – Cisco Proprietary Skinny Control Protocol Communicates with Cisco SCCP Equipment OSS – Open Sound System Older Linux Sound Drivers Communicates with Soundcards ALSA – Advanced Linux Sound Architecture New Linux Sound Drivers Communicated with Soundcards

34. The Codec Translation API Codec Translation API Converts Audio Codecs G.711 Ulaw/Alaw Ulaw is used in the states, Alaw in Europe G.726 32Kbps G.729 Requires a license ($10 / channel from Digium) Most widely deployed, low bandwidth codec (8kbps) GSM iLBC LPC10 (not recommended!) Speex Open Source, Royalty Free, configurable 4-48kbps, VBR, ABR

35. The File Format API This API Allows Reading/Writing of Various File Formats Some applications may need to archive digital audio streams in different formats Used by many applications such as VoiceMail, which records messages to disk in whatever format you choose Available Formats WAV MP3 AU GSM

36. The Application API Applications Perform Functions Modules of code that are used by the Dial Plan For Example: Answer: Answer a channel if ringing BackGround: Play a file while awaiting DTMF tones Busy: Indicate busy condition (normal busy) Congestion: Indicate congestion (fast busy) Dial: Place a call and connect to the current channel Directory: Provide directory of voicemail extensions MeetMe: MeetMe conference bridge MP3Player: Play an MP3 file or stream MusicOnHold: Play Music On Hold indefinitely Record: Record to a file VoiceMail: Leave a voicemail message VoiceMailMain: Enter voicemail system

37. The Application API Simple Dial Plan Example Dial 216-920-3111 w/ your Cell Phone to hear it live ; CallerID Identify exten => 2169203111,1,Answer exten => 2169203111,2,Wait(2) exten => 2169203111,3,Playback(channel-insecure-warn) exten => 2169203111,4,SayDigits(${CALLERIDNUM}) exten => 2169203111,5,Wait(1) exten => 2169203111,6,SayDigits(${CALLERIDNUM}) exten => 2169203111,7,Wait(1) exten => 2169203111,8,Playback(goodbye) exten => 2169203111,9,Hangup

38. Console Output -- Executing Answer("Zap/1-1", "") in new stack -- Accepting call from '2164114184' to '2169203111' on channel 0/1, span 1 -- Executing Wait("Zap/1-1", "2") in new stack -- Executing Playback("Zap/1-1", "channel-insecure-warn") in new stack -- Playing 'channel-insecure-warn' (language 'en') -- Executing SayDigits("Zap/1-1", "2164114184") in new stack -- Executing Wait("Zap/1-1", "1") in new stack -- Executing SayDigits("Zap/1-1", "2164114184") in new stack -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Playback("Zap/1-1", "goodbye") in new stack -- Playing 'goodbye' (language 'en') -- Executing Hangup("Zap/1-1", "") in new stack == Spawn extension (inbound, 2169203111, 9) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' asterisk*CLI>

39. API Access C API Accessible via standard ANSI C Pre-existing example code for applications, channel drivers etc.. Forms the Core of Asterisk Well documented, just read the code ;)

40. res_perl Similar to mod_perl for Apache Single Perl interpreter is loaded and used to process requests Allows embedding of perl commands directly in Dial Plan For the more adventurous, can be used to extend Asterisk to unimaginable tasks Available as part of the asterisk_addons package from CVS

41. res_js Similar to res_perl, except for Javascript Available from http://www.pbxfreeware.org

42. AGI Asterisk Gateway Interface Similar to CGI Write in whatever you want (Perl, PHP, Python, Pascal, Java, BASH… ) Variables are passed on StdIn to your Applications, results and commands are passed back on StdOut Included w/ Asterisk, no additional work required

43. Manager API Allows client/server interaction over TCP/IP sockets w/ authentication Can be used to issue commands or monitor PBX events Used by applications such as the Flash Operator Panel and IP Switchboard

44. Pre-Recorded Prompts Hundreds of professionally recorded prompts Recorded by Allison Smith, a Voice Over Professional http://www.theivrvoice.com Clients include; Target Bell Canada Volkswagen Cingular Victoria Secret Can do custom work hourly or using a credit system For more information visit: http://thevoice.digium.com

45. Connecting Asterisk To The World Many, Many Options.. TDM Cards from Digium VoIP Softphones VoIP Hardware from Various Vendors VoIP Termination / Origination Service from Carriers The “ITSP” Internet Telephony Service Provider

46. X100P TDM400 TDM Hardware from Digium

47. T100P TE405P TDM Hardware from Digium

48. DS3000P S100 IAXY TDM Hardware from Digium

49. Linksys PAP-NA2 Analog Telephony Adapters

50. SIP Hardware Phones Cisco 7960 Polycom IP-600