Asterisk. Michael Kershaw

Transcription

1 Asterisk *

2 What Asterisk Can Do Voice Over IP (VOIP) Physical phone switch (PBX) Software phone switch Answering machine Call trees (Press 1 to...)

3 VOIP Voice Over IP: Make telephone calls using the Internet. Cheap (real phone calls) or free (VOIP to VOIP) Commercial/Semi-Commercial services like Skype, Vonage. Like compressing a CD to MP3 files, the goal is to make the voice data as small as possible without losing quality. Software, hardware, and standard phone converters.

4 VOIP and Asterisk Asterisk supports several VOIP speech compression codecs. Will not work with commercial VOIP solutions like Vonage or Skype (different networks) Asterisk can route VOIP calls: Over the Internet to other Asterisk users Over the Internet to VOIP gateway providers Into the real phone network, if you have the right hardware

5 POTS And Asterisk POTS Plain Old Telephone System. Requires special hardware: FXS and FXO. FXS emulates the phone system. You can plug a normal phone into a FXS adapter. FXO emulates a phone device. You can plug a PC into the POTS network with a FXO adapter. High-end 'T' line interface hardware. Similar to FXO but can trunk dozens of lines over a T1 carrier.

6 Talking To Asterisk VOIP or real phone, you need a way to talk to Asterisk. Analog phone to VOIP adapters: Ethernet on one side, telephone jack on the other, lets you use any existing phone. Usually talk SIP protocol. FXS: Acts like a sound card in your Asterisk server, lets you use any existing phone. Hardware VOIP phone: Looks like a real phone, but talks SIP to Ethernet. Software VOIP phone: Run it on your computer.

7 Installing Asterisk Download Asterisk from or look in your distribution for a package. Extract the source, compile, and install the compiled code. Asterisk doesn't use the standard autoconfig tools, so a./configure is not needed. tar xvf asterisk tar.gz make make install

8 Asterisk Configuration Files System configurations live in /etc/asterisk. There are a lot of them. The most important ones are: sip.conf: controls SIP VOIP users. iax.conf: controls IAX VOIP connections. extensions.conf: controls phone numbers and call trees, both incoming and outgoing calls. voic .conf: controls voice mail boxes, passwords, etc.

9 Configuring Asterisk For SIP Most VOIP stuff uses SIP (software phones, hardware adapters, etc) Needs a SIP configuration entry in sip.conf for each SIP device. SIP entry defines type, extension context, name, password, voic mailbox, and audio codecs allowed.

10 Example SIP User [dragorn grandstream] type=friend context=nycccp username=dragorn grandstream fromuser=dragorn grandstream secret=monkey callerid=dragorn <1020> host=dynamic nat=no canreinvite=yes dtmfmode=info disallow=all allow=ulaw

11 Things To Know About SIP Extremely basic protocol. Stream of UDP packets containing voice data. CAN pass through NAT, but doesn't do so very well in all situations. Container around multiple audio codecs: ulaw, alaw, gsm, and more. Different codecs have different speech quality. Not all codecs can handle DTMF (touchtone).

12 SIP Devices Assorted software phones: kphone, linphone, mythphone, gnophone to name a few. Look on Hardware phones: Budgetone is cheap & decent Standalone FXS to SIP adapters

13 Asterisk Extensions Configuration Very complex file, but important for operation. Boring part of the presentation, but controls all the neat things Asterisk can do. Config is divided into segments, aka calling domains. Each extension can perform multiple actions: Dialing another extension Playing background audio Dial single or multiple local phones Execute a call tree or macro

14 * Extensions: Outbound Dialing Wildcard extension matches outbound numbers (1-XXX-XXX-XXXX) Set our caller ID to (hah!) Direct it to the external provider via the IAX2 protocol (Asterisk to Asterisk protocol, supported by our example outbound dialing provider, and configured in iax.conf) exten => _1NXXNXXXXXX,1,SetCallerID( ) exten => _1NXXNXXXXXX,2,Dial,IAX2/foo/${EXTEN}

15 * Extensions: Local Phones Define a macro which calls the SIP address of the specified phone, waits a maximum of 20 seconds, and then directs the call based on the status. Set an extension in our local phone domain and link it to the SIP phone. [macro sipexten] exten => s,1,dial(sip/${arg1},20) exten => s,2,goto(s ${DIALSTATUS},1) [nycccp phones] exten => 1020,1,Macro(sipexten,dragorn grandstream)

16 * Extensions: Calling Trees Include local phone domains. Set up background speech. Script the call tree and fall through to calling local users if nothing else is entered. [nycccp incoming] include => nycccp_phones include => nycccp vmdefaults exten => s,1,background(nycccp intro) exten => 2,1,Goto(nycccp departments,s,1) exten => 3,1,Goto(nycccp orginizations,s,1) exten => 4,1,Goto(nycccp people,s,1) exten => 11,1,Goto(nycccp toolkit,s,1) exten => 2,1,Dial(SIP/dragorn grandstream&sip/porkchop,10t)

17 * Extensions: CID Spoofing CID spoofing is trivial, depending on your VOIP outbound provider. Match #*# and set the CID to the first number. Your provider may get unhappy with you doing this a lot! exten => _1NXXNXXXXXX*NXXNXXXXXX,1, SetCallerID(${EXTEN: 10}) exten => _1NXXNXXXXXX*NXXNXXXXXX,2, Dial,IAX2/foo/${EXTEN:0:11}

18 Voic Builtin voic system. Per-extension mailbox and passwords. Incoming messages recorded as wav, mp3. Remotely checkable if you add the voic system to your dialable extensions. notification of new messages.

19 Voice Synthesis Asterisk can use realtime voice synthesis via Festival. Unfortunately, Festival sucks to compile and install the extra voices. Voice quality is passable but not great. Often better to record your own messages. Requires Festival daemon to be running.

20 IAX, Asterisk Protocol Two Asterisk installations can communicate directly with each other. Multiple phone switches in one location can share the load. Direct connections between your house and others via the Internet: Free calls to your friends! Some clients talk IAX directly (gnophone). More robust than SIP but less widespread.

21 Going From VOIP To World Free World Dialup provides Free (imagine that) numbers to reach other FWD members. VoipJet provides outbound connections to the real phone network at very cheap rates ($0.001/min inside the US and $0.03/min average international). Stanaphone provides NY inbound phone numbers. You don't have to use the same inbound and outbound providers.

22 VoipJet Is Weird The first rule about VoipJet is you do not talk about VoipJet. In many ways they seem very sketchy. EULA prohibits talking about using them or disclosing that you use them. Service is still good and you can pay via PayPal. Weirdness is probably to avoid being taxed as a telecom provider.

23 Business-Level Asterisk T1 interface card. Use Asterisk as a real phone PBX to trunk multiple extensions to the T1 line. VOIP to the desk, or FXS to standard phones. Much cheaper to run than a commercial PBX system. Uses the ZapTel kernel modules to drive analog and digital phone interface cards. Never done it personally.

24 Tricks With Asterisk + POTS Using a FXO card, connects to your home telephone jack. Route incoming calls to a call tree. Amaze your friends, annoy the telemarketers. Route outgoing calls over phone or VOIP depending on area code to save money. Digital answering machine. MP3s of your incoming messages.

25 VOIP Legalisms Commercial VOIP providers are likely to be treated as telecom providers (= taxes!) FCC/Feds are planning to require VOIP providers make it wiretappable. (= spying!) 911 support becoming a major issue. How does the 911 system know where you are? None of this applies to you at home building your own, but they may apply to the providers you get inbound/outbound service from. Primarily targets Vonage, etc.

26 Stuff To Worry About If your only phone is VOIP (Asterisk or commercial), what happens if your power goes out or your Internet link goes down? VOIP is really too high bandwidth for dialup. Broadband of some sort is necessary. VOIP is sniffable. It's not legal, but that won't stop it. Your phone is also tappable. Take your pick.

27 VOIP Quality VOIP speech quality is directly affected by the available bandwidth. Stuttering and dropouts will happen if your line isn't fast enough. Packet shaping can help if your line is congested by uploads and downloads. If you use Linux for packet shaping, a PFIFO Queue is most appropriate: The short packet queue prevents VOIP data from backing up causing stuttering. Shaping scripts like WonderShaper can help automate this.

28 Asterisk Config Helpers has links to many Asterisk frontends and utilities LiveCD distros and specific modified distro installs such as Xorcom Several PHP configuration and billing frontends has live diagnostics and monitoring

29 Asterisk Flash Operator Panel

30 Outro Asterisk: Incoming: Outgoing: Info & Wiki: More Software: Us: