Asterisk SIP PBX simple tutorial / quick start guide

Recently I start to investigate how to make asterisk to be an SIP BPX with small foot print, and I have a running SIP PBX now. Below are the notes on how I got it to run.

Platform: I am running asterisk in Colinux under Windows Vista. Debian 5 is running in Colinux.

Short summary:

version: asterisk 1.4 is stable and used widely. 1.6 is considered short-term support. Supposedly 1.8 is another stable version for long-term support. I use 1.4
source : the source tar gzip of asterisk is about 23MB. It uses the standard "./configure;make;make install" procedure to compile. See README in source tar ball.
structure: asterisk uses a lot of ".so" dynamic libraries, which are called modules and are loaded dynamically when program starts. Which one to load or not to load is controlled by the file "modules.conf". Many of the modules are essential to make asterisk useful, while others are optional for our purpose.
directories: configurations are under /etc/asterisk, modules (dynamic library files are under /usr/lib/asterisk/modules). Other directories are determined in compile-time and are listed in "asterisk.conf"
configurations: Unlike most unix programs, "asterisk.conf" is not what you change the most. In fact, you can probably leave it as is. The files we need to change the most for making a IP PBX are:

• modules.conf ; for configuring which modules to load or not load
• sip.conf ; for configuring all sip channels, both external and internal
• extensions.conf; the heart of the PBX, configures what key press/ what extension does what

1. apt-get install asterisk
2. /etc/init.d/asterisk stop.  I like to use console for getting thing to run. so stop the daemon
3. copy modules.conf below as your modules.conf
4. copy sip.conf as your sip.conf. I use sipgate as my provider. ( I tested incoming call and outgoing call to toll-free numbers)
5. copy extensions.conf to yours.
6. start your asterisk in console mode (asterisk -cvvv)
7. install x-lite software phone on your Windows and configure it as follows:

8. now you can dial 123 to hear the playback voice from asterisk. go to asterisk CLI, and type "sip show peers" and you should see two peers, your sipgate and your x-lite phone.
9. You can now make calls and receive calls. 
10. For further reading, I recommend the O'reilly book "Asterisk".

modules.con

[modules]
autoload=yes
noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so
noload => chan_modem.so
noload => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => pbx_dundi.so
noload => pbx_realtime.so
noload => app_directory.so
noload => app_userevent.so
noload => app_voicemail.so
noload => app_voicemail_imap.so
noload => app_voicemail_odbc.so
noload => pbx_ael.so
noload => app_directory_odbc.so
noload => app_zapateller.so
noload => app_zapbarge.so
noload => app_zapras.so
noload => app_zapscan.so
noload => cdr_custom.so
noload => cdr_manager.so
noload => cdr_odbc.so
noload => cdr_pgsql.so
noload => cdr_radius.so
noload => cdr_sqlite.so
noload => chan_agent.so
noload => chan_alsa.so
noload => chan_gtalk.so
noload => chan_iax2.so
noload => chan_mgcp.so
noload => chan_oss.so
noload => chan_phone.so
noload => chan_vpb.so
noload => chan_zap.so
noload => codec_zap.so
noload => format_h264.so
noload => format_jpeg.so
noload => format_mp3.so
noload => format_ogg_vorbis.so
noload => pbx_ael.so
noload => pbx_dundi.so
noload => pbx_loopback.so
noload => pbx_realtime.so
noload => pbx_spool.so
noload => res_config_odbc.so
noload => res_config_pgsql.so
noload => res_jabber.so
noload => res_odbc.so
noload => res_smdi.so
noload => res_snmp.so
noload => res_speech.so
noload => res_watchdog.so

[global]

sip.conf

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

register => YOUR-SIP-ID:YOUR-SIP-PASSWD@sipgate/YOUR-SIP-ID

[sipgate]
type=peer
secret=YOUR-SIP-PASSWD
insecure=invite
username=YOUR-SIP-ID
defaultuser=YOUR-SIP-ID
fromuser=YOUR-SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
allow=ilbc
allow=g729
dtmfmode=rfc2833
nat=yes

[1000]
type=friend
context=phones
host=dynamic
qualify=yes

extensions.conf

[general]

[globals]

[sipgate_in]
exten => YOUR-SIP-ID,1,Dial(SIP/1000,30,r)
exten => YOUR-SIP-ID,n,Hangup

[sipgate_out]
exten => _X.,1,Set(CALLERID(num)=YOUR-SIP-ID)
exten => _X.,n,Dial(SIP/${EXTEN}@sipgate,30,trg)
exten => _X.,n,Hangup

[phones]
exten => 123,1,Answer()
exten => 123,n,Background(demo-congrats)
exten => 123,n,WaitExten()

include => outbound-long-distance

exten => 2,1,Playback(digits/2)
exten => 2,n,Goto(phones,123,1)

exten => 3,1,Playback(digits/3)
exten => 3,n,Goto(phones,123,1)

exten => i,1,Playback(pbx-invalid)
exten => i,n,Goto(123,1)

exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()

[outbound-long-distance]
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@sipgate,30,trg)
exten => _91NXXNXXXXXX,n,Playtones(congestion)
exten => _91NXXNXXXXXX,n,Hangup()