Debian: Asterisk PBX
From ReceptiveIT
Contents |
Overview
Asterisk is an Open Source software based PBX that runs on Linux, BSD and MacOSX. It provides all the features that you would expect from a real hardware phone system, with a flexibility that you probably never thought possible. In short, Asterisk is the holy grail of telephony. Many thanks go out to all developers, and to Digium for making Asterisk a reality.
Installation
The instructions provided here are for Debian GNU/Linux. These instructions might translate to other distributions of Linux.
Install Asterisk
apt-get install asterisk
Was that it? It can't be that easy...
Configuration
For a phone system to be useful, we need to have at least one handset (extension) and one incoming line. This is what is called a 1x1 PBX. In the real world, a 1x1 PBX would be a little overkill, but as an example of how Asterisk works, it is an easily attainable first step.
Configuring your first handset
This example will assume that you have a usable SIP phone to use as a handset for Asterisk. This SIP phone could be a Ethernet based SIP phone, a Ethernet to POTS ATA Device, or a software SIP phone running as an application on a PC. Edit /etc/asterisk/sip.conf and add the following at the bottom of the file
[101] type=friend context=default secret=password callerid="First Handset" <101> host=dynamic disallow=all allow=g729 allow=gsm allow=ulaw
Edit /etc/asterisk/extensions.conf and add the folling just after [default]
exten => 101,1,Macro(stdexten,101,SIP/101,80,Ttr)
You should now be able to dial numbers from your SIP phone. You can log into Asterisk from the command line and see events happen in real-time.
asterisk -vvvr
Zaptel Hardware
zaptel.conf - Don't forget that FXO ports are signalled with FXS, and FXS ports are signalled with FXO! The following example sets up a four (4) channel analog FXS card with Australian dial tone.
# # Zaptel Configuration File # loadzone = au defaultzone=au fxoks=1-4
Simply setting the country code does not set the correct impedance for the line. By default, the TDM400 hardware sets itself to FCC mode, which is North American lines. If you live in Australia, like me, you will need to edit /etc/modprobe.d/zaptel and add the following line.
options wctdm opermode=AUSTRALIA
Dial Plan
Linksys (Sipura) SPA-3000
Features - 1 x FXO port and 1 x FXS port
This little box has proven to be a good, reliable and cheap way to get a single line into your Asterisk box. While the Digium cards are fantastic, sometimes you just need a single line.
Setting up is quite easy. Log into your SPA-3000 as admin, and in advanced mode.
Regional
Call Progress Tones FXS Port Impedance: 220+820||120nF FXS Port Input Gain: 3 FXS Port Output Gain: -3 Detect ABCD: yes Playback ABCD: yes Caller ID Method: Bellcore(N.Amer,China) FXS Port Power Limit: 3
PSTN Line
Line Enable: yes Proxy and Registration Proxy: {IP address of Asterisk Server} Use Outbound Proxy: yes Outbound Proxy: {IP address of Asterisk Server} Use OB Proxy in Dialog: yes Register: yes Make Call Without Reg: no Register Expires: 3600 Ans Call Without Reg: no Use DNS SRV: no DNS SRV Auto Prefix: no Proxy Fallback Intvl: 3600 Subscriber Information Display Name: {SIP Username} User ID: {SIP Username} Password: {SIP Password} Use Auth ID: no Audio Configuration Preferred Codec: G711u Dial Plans Dial Plan 1: (xx.) Dial Plan 2: (S0<:192.168.198.23>) VoIP-To-PSTN Gateway Setup VoIP-To-PSTN Gateway Enable: yes VoIP Caller Auth Method: none VoIP PIN Max Retry: 3 One Stage Dialing: yes Line 1 VoIP Caller DP: 1 VoIP Caller Default DP: 1 Line 1 Fallback DP: none PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN CAller Auth Method: none PSTN Ring Thru Line 1: yes PSTN PIN Max Retry: 3 PSTN CID For VoIP CID: no PSTN Caller Default DP: 2 FXO Timer Values (sec PSTN Answer Delay: 0 PSTN Disconnect Detection Detect CPC: yes Detect Polarity Reversal: yes Detect PSTN Long Silence: no Detect VoIP Long Silence: no PSTN Long Silence Duration: 30 VoIP Long Silence Duration: 30 PSTN Silence Threshold: medium Min CPC Duration: 0.2 Detect Disconnect Tone: yes Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2) International Control FXO Port Impedance: 220+820||120nF Ring Frequency Min: 10 SPA To PSTN Gain: 0 Ring Frequency Max: 100 PSTN To SPA Gain: 0 Ring Validation Time: 256ms Tip/Ring Voltage Adjust: 3.1V Ring Indication Delay: 512ms Operational Loop Current Min: 16mA Ring Timeout: 640ms On-Hook Speed: 26ms (Australia) Ring Threshold: 13.5-16.5 Vrms Current Limiting Enable: no Ringer Impedance: High (Normal) Line-In-Use Voltage: 30
We also have to add an entry into the sip.conf file
[pstn] type=friend host=dynamic secret=password call-limit=2 disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 allow=gsm context=pstnincoming [4350] type=friend context=sipuser secret=password auth=md5 callerid="Cordless" <4350> host=dynamic nat=yes dtmfmode=rfc2833 canreinvite=no reinvite=no dtmfmode=info incominglimit=2 mailbox=4350 disallow=all allow=g729 allow=gsm allow=ulaw
Next we need to add the PSTN line to the correct context, and the FXS port to the sip phones context, in the extensions.conf file.
[pstnincoming] # What incoming calls from the PSTN have access to exten => s,1,Dial(SIP/4351&SIP/4350|30,tr) exten => s,2,Voicemail(u4351) [sipuser] # What VoIP users have access to include => voip-phones include => tollcalls [voip-phones] # VoIP Extensions exten => 4350,1,Macro(stdexten,4350,SIP/4350,80,Ttr) exten => 4350,2,Voicemail(u4350)
BLF and Line Keys
Linksys Handsets
Linksys SPA942 can now do BLF and speeddials really well
Phone Menu - Line Key Extended Function
Asterisk
Phone Menu - Extended Function
fnc=blf+sd+cp;sub=848@$PROXY;ext=848@$PROXY;vid=1
Asterisk Config
sip.conf
subscribecontext = handsets notifyringing = yes notifyhold = yes
Each handsets need a call limit
call-limit=1
Inbound context
[pstn-incoming]
exten => 62160812,1,NoOp(Reception Test)
exten => 62160812,n,Set(TIMEOUT(digit)=2)
exten => 62160812,n,Set(SWITCHMODE=${DB(cvh/nightswitch)})
exten => 62160812,n,GotoIf($["${SWITCHMODE}" = "0"]?normal)
exten => 62160812,n,GotoIf($["${SWITCHMODE}" = "1"]?meeting)
exten => 62160812,n,GotoIf($["${SWITCHMODE}" = "2"]?morning)
exten => 62160812,n,GotoIf($["${SWITCHMODE}" = "3"]?night)
exten => 62160812,n,Goto(hangup)
exten => 62160812,n(meeting),NoOp(Meeting)
exten => 62160812,n,Playback(custom/ceh-meetingmsg)
exten => 62160812,n,Goto(hangup)
exten => 62160812,n(morning),NoOp(Morning)
exten => 62160812,n,Playback(custom/ceh-morningmsg)
exten => 62160812,n,Goto(hangup)
exten => 62160812,n(night),NoOp(Night)
exten => 62160812,n,Playback(custom/ceh-ahmsg)
exten => 62160812,n,Goto(hangup)
exten => 62160812,n(normal),NoOp(Business Hours)
exten => 62160812,n,Macro(directdial,812,1)
exten => 62160812,n,Goto(hangup)
exten => 62160812,n(hangup),NoOp(Hanging up)
exten => 62160812,n,Hangup
BLF Context
[ceh-callroute]
exten => 848,hint,Custom:ceh
exten => 848,1,NoOp(CEH Nightswitch)
exten => 848,n,Set(SWITCHMODE=${DB(ceh/nightswitch)})
exten => 848,n,Background(custom/press&custom/0&custom/2&custom/activate&custom/daymode,m)
exten => 848,n,Background(custom/press&custom/1&custom/2&custom/activate&custom/meetingmode,m)
exten => 848,n,Background(custom/press&custom/2&custom/2&custom/activate&custom/morningmode,m)
exten => 848,n,Background(custom/press&custom/3&custom/2&custom/activate&custom/nightmode,m)
exten => 0,1,NoOp(Set Day Mode)
exten => 0,n,Set(DB(ceh/nightswitch)=0)
exten => 0,n,Set(DEVICE_STATE(Custom:ceh)=NOT_INUSE)
exten => 0,n,Playback(custom/daymode&custom/hasbeenset)
exten => 0,n,Hangup()
exten => 1,1,NoOp(Set Meeting Mode)
exten => 1,n,Set(DB(ceh/nightswitch)=1)
exten => 1,n,Set(DEVICE_STATE(Custom:ceh)=INUSE)
exten => 1,n,Playback(custom/meetingmode&custom/hasbeenset)
exten => 1,n,Hangup()
exten => 2,1,NoOp(Set Morning Mode)
exten => 2,n,Set(DB(ceh/nightswitch)=2)
exten => 2,n,Set(DEVICE_STATE(Custom:ceh)=INUSE)
exten => 2,n,Playback(custom/morningmode&custom/hasbeenset)
exten => 2,n,Hangup()
exten => 3,1,NoOp(Set Night Mode)
exten => 3,n,Set(DB(ceh/nightswitch)=3)
exten => 3,n,Set(DEVICE_STATE(Custom:ceh)=INUSE)
exten => 3,n,Playback(custom/nightmode&custom/hasbeenset)
exten => 3,n,Hangup()
Receiving Faxes
There is a cool little utility called IAXModem which is a software modem that creates analog tones digitally...
IAXModem becomes an IAX client to asterisk. When a call comes in on the fax line, asterisk passes the call to IAXModem, which does the software modem thing, and then communicates to Hylafax over a tty that IAXModem generates.
Asterisk Configuration
Edit iax.conf and add the following
[299] type=friend host=dynamic context=incoming-faxes jitterbuffer=no disallow=all allow=slin secret=iaxmodem auth=md5 qualify=yes
IAXModem Configuration
Edit /etc/iaxmodem/iaxmodem-cfg.ttyIAX0 and enter the following
device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 refresh 300 server 127.0.0.1 peername 299 secret iaxmodem cidname Business Name cidnumber 61262001122 codec slinear
Hylafax Configuration
Adding a Fax Modem
Here is a sample /etc/hylafax/config.ttyIAX0
# $Id: iaxmodem,v 1.1 2006/04/02 23:18:30 darren Exp $ # # prototype config for the IAXmodem softmodem which uses # the spandsp soft-DSP library # # CountryCode: 61 AreaCode: 2 FAXNumber: "62001122" LongDistancePrefix: 0 InternationalPrefix: 0011 DialStringRules: etc/dialrules ServerTracing: 0xFFF SessionTracing: 0xFFF RecvFileMode: 0600 LogFileMode: 0600 DeviceMode: 0600 RingsBeforeAnswer: 1 SpeakerVolume: off GettyArgs: "-h %l dx_%s" LocalIdentifier: "Business Name" TagLineFont: etc/lutRS18.pcf TagLineFormat: "From %%l|%c|Page %%P of %%T" MaxRecvPages: 200 ModemReadyState: D # # # # Modem-related stuff: should reflect modem command interface # and hardware connection/cabling (e.g. flow control). # ModemType: Class1 # use this to supply a hint # # The modem is taken off-hook during initialization, and then # placed back on-hook when done to prevent glare. # ModemResetCmds: "ATH1\nAT+VCID=1" # enables CallID display ModemReadyCmds: ATH0 Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response Class1RMQueryCmd: "!24,48,72,96" # V.17 fast-train recv doesn't work well CallIDPattern: "NMBR=" CallIDPattern: "NAME=" CallIDPattern: "ANID=" CallIDPattern: "NDID=" # Uncomment these if you really want them, but you probably don't. #CallIDPattern: "DATE=" #CallIDPattern: "TIME="
Emailing Faxes
You need to add a file called /etc/hylafax/FaxDispatch. Here is a simple example.
FILETYPE=pdf; SENDTO=FaxMaster; MIMENCODE=bin/uuencode_it;
Here is a more comprehensive example
## $Id: FaxDispatch,v 1.2 2003/05/04 23:49:41 darren Exp $
##
## Default FaxDispatch file - routes all inbound faxes to FaxMaster as PDF
##
## Consult the faxrcvd(8C) man page for more information
##
#SENDTO=faxMaster; # by default email to FaxMaster
#FILETYPE=pdf; # in PDF format
##
## This excerpt from the man page gives you an idea of what's possible here
##
## You can route by sender's TSI
#case "$SENDER" in
# *1*510*526*1212*) SENDTO=sam;; # Sam's test rig in Berkeley
# *1*415*390*1212*) SENDTO=raster@asd;; # 7L Xerox room, used for scanning
# *5107811212) SENDTO=peebles@mti;; # stuff from home
#esac
## and/or by device
#case "$DEVICE" in
# ttyS1) SENDTO=john;; # all faxes received on ttyS1
# ttyLT0) SENDTO=mary@home;; # all faxes received on ttyLT0
#esac
## and/or by device
FOLDER="/var/spool/hylafax/recvq/"
FULLPATH="${FOLDER}${FILENAME}.tif"
case "$DEVICE" in
ttyS14) mv $FULLPATH /var/spool/hylafax/recvq/dept1/;; # all faxes received on ttyS14
ttyS15) mv $FULLPATH /var/spool/hylafax/recvq/dept2/;; # all faxes received on ttyS15
esac
## and/or by caller id
#case "$CIDNUMBER" in
# 435*) SENDTO=lee; FILETYPE=pdf;; # all faxes from area code 435
# 5059627777) SENDTO=amy; FILETYPE=tif;; # Amy wants faxes in TIFF
#esac
Recording Calls
Note: Recording calls may or may not be legal in your area. If you decide to record your calls, make sure you are legally in a position to do so.
The following code, inserted into extensions.conf will start recording a call if the number starts with 8.
;
; Record calls
;
exten => _8.,1,Set(DIREC=/var/spool/asterisk/voicerecord)
exten => _8.,2,Set(CALLFILENAME=${TIMESTAMP}-${CALLERIDNUM}-${EXTEN:1})
exten => _8.,3,MixMonitor(${DIREC}/${CALLFILENAME}.wav|av(0)V(0))
exten => _8.,4,Dial(${TRUNK}/${EXTEN:1}|60,t)
exten => _8.,5,Congestion
It would be a good idea to schedule a cron job to convert the wav files to ogg or mp3 after hours to conserve disk space.
CDR using PostgreSQL
apt-get install postgresql-8.1
edit /etc/asterisk/cdr_pgsql
[global] hostname=localhost port=5432 dbname=asterisk password=password user=postgres
Switch User to the Postgres user
su postgres
Create the database
createdb asterisk
Create the CDR table
psql asterisk < /usr/share/doc/asterisk-doc/contrib/scripts/postgres_cdr.sql
Create the user
createuser -A -D -P asterisk Enter password for new user: Enter it again: CREATE USER
Log into PostgreSQL
psql asterisk
Grant access to the user
grant INSERT on cdr to asterisk; grant SELECT on cdr to asterisk; grant ALL on cdr_acctid_seq to asterisk;
edit /etc/postgresql/8.1/main/pg_hba.conf
local all postgres ident sameuser # TYPE DATABASE USER CIDR-ADDRESS METHOD # "local" is for Unix domain socket connections only local all all md5 sameuser # IPv4 local connections: host all all 127.0.0.1/32 md5 # IPv6 local connections: host all all ::1/128 md5
Music On Hold
Here is a script to convert mp3 files to wav for non-processor intensive music on hold.
#!/bin/bash FROMDIR="/root/mp3" TODIR="/var/lib/asterisk/moh" for i in $FROMDIR/*.mp3 do BASEFILE=$(basename $i .mp3) echo Converting $BASEFILE mpg123 -s --rate 44100 --mono $FROMDIR/$BASEFILE.mp3 > $TODIR/$BASEFILE.raw sox -r 44100 -2 -s -c 1 $TODIR/$BASEFILE.raw -r 8000 -c 1 $TODIR/$BASEFILE.wav #sox $TODIR/$BASEFILE.wav -t gsm -r 8000 -b -c 1 $TODIR/$BASEFILE.gsm #sox $TODIR/$BASEFILE.wav -t ul -r 8000 -b -c 1 $TODIR/$BASEFILE.pcm rm -f $TODIR/$BASEFILE.raw done
/etc/asterisk/musiconhold.conf
[default] mode=files directory=/var/lib/asterisk/moh
