Step 1.2 – DAHDI installation
Dahdi est un logiciel anciennement appelé zaptel qui fournit une interface pour pouvoir faire communiquer asterisk avec les cartes de communication telephonique (analogique ou numérique). Dahdi est indispensable pour certaine fonctionnalité d’asterisk comme le trunking IAX ou conférence meetme.
On se place dans le répertoire source : cd/usr/src
La commande ls permet de visionner le contenu du repertoire
On se place dans le répertoire de dahdi : cd dahdi-linux-complete-2.4.0+2.4.0
On relance une commande ls pour en voire le contenu
(LESS README permet de voir le readme)
1ère étape avec les droit # : on fait un make all
2ème étape : make install
3ème étape : make config
4ème étape : /etc/init.d/dahdi start
Step 1.4 – libpri compilation
La libpri est une librairie utilisée pour les cartes de téléphonie numérique les BRI et PRI.
Meme si nous n’avons pas de carte de communication, il faut les installer car c’est encore une fois un pré-requis à l’installation d’Asterisk .
1ère étape : cd usr/src/libpri-1.4 …
2ème étape : make
3ème étape : make install
Step 1.5 – Asterisk compilation
1ère étape : cd /usr/src/asterisk-1.8.3-rc3/
2ème étape :./configure
3ème étape : make
4ème étape : make install
5ème étape : make samples
6ème étape : make config
Step 1.6 – Creating a simple DialPlan
/etc/asterisk/sip.conf
pout vider le fichier :
cat /dev/null > sip.conf
nano sip.conf
[general]
bindaddr=0.0.0.0
bindport=5060
disallow=all
allow=ulaw
context=internal_calls
dtmfmode=rfc2833
allowoverlap=no
[500]
defaultuser=500
type=friend
qualify=yes
nat=no
secret=1234
host=dynamic
directmedia=yes
[501]
defaultuser=501
type=friend
port=5061
qualify=yes
nat=no
secret=4321
host=dynamic
directmedia=yes
Dans /etc/asterisk/extensions.conf
Nano extensions.conf
[internal_calls]
exten => 500, 1, Answer()
exten => 500, 2, Dial(SIP/500)
exten => 500, 3, Hangup()
exten => 501, 1, Answer()
exten => 501, 2, Dial(SIP/501)
exten => 501, 3, Hangup()
Step 1.7 – Implementing the Toulouse IPBX
sip-paris.society.lan
Fichier /etc/asterisk/iax.conf
[general]
bandwidth=low
disallow=lpc10
bindaddr=192.168.143.134
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
[asterisk2]
type=friend
defaultuser=asterisk2
secret=password
auth=plaintext
context=internal_calls
host=dynamic
File /etc/asterisk/extensions.conf
[calls_to_toulouse]
exten => 600,1,Dial(IAX2/asterisk2/600)
exten => 601,1,Dial(IAX2/asterisk2/601)
[internal_calls]
include => calls_to_toulouse
(Ö)
sip-toulouse.society.lan
Fichier /etc/asterisk/iax.conf
[general]
bandwidth=low
disallow=lpc10
bindaddr=192.168.143.133
jitterbuffer=no
forcebitterbuffer=no
autokill=yes
register=>asterisk2:password@192.168.198.134
[asterisk2]
type=friend
defaultuser=asterisk2
secret=strong_password
auth=plaintext
context=internal_calls
host=192.168.198.134
trunk=yes
File /etc/asterisk/extensions.conf
[calls_to_paris]
exten => 500,1,Dial(IAX2/asterisk2/500)
exten => 501,1,Dial(IAX2/asterisk2/501)
[internal_calls]
include => calls_to_paris
(Ö)
Step 2.1 – Audio conference
On each IPBX, in the file /etc/asterisk/extensions.conf
[conference_room]
exten => 900, 1, Answer()
exten => 900, 2, ConfBridge(1,scM)
[internal_calls]
include => conference_room
(Ö)
Step 2.2 – Voicemail
#sip-paris.society.lan
File /etc/asterisk/voicemail.conf
[voicemail]
500 => 5678,500
501 => 8765,501
File /etc/asterisk/sip.conf
[500]
(Ö)
mailbox=500@voicemail
[501]
(Ö)
mailbox=501@voicemail
in the file /etc/asterisk/extensions.conf
[local_voicemail]
exten => 777,1,Answer()
exten => 777,2,VoiceMailMain(@voicemail)
exten => 777,3,HangUp()
[internal_calls]
(…)
exten => 500,1,Answer()
exten => 500,2,Dial(SIP/500,20)
exten => 500,3,VoiceMail(500@voicemail)
exten => 500,4,PlayBack(vm-goodbye)
exten => 500,5,Hangup()
include=>local_voicemail
Step 2.3 – Speaking Clock
/etc/asterisk/extensions.conf
[speaking_clock]
exten => 3669,1,Answer()
exten => 3669,2,SayUnixTime(,Europe/Paris,AdBY kM)
exten => 3669,3,HangUp()
And
[internal_calls]
include => speaking_clock
(…)
Step 2.4 - Interactive menu
Definition de IVR : Interactiv Voice Responce
[IVR]
exten => 1337,1,Read(digit,hello-world,1)
exten => 1337,2,GotoIf($["${digit}" = “1”]?internal_calls,3669,1)
exten => 1337,3,GotoIf($["${digit}" = “2”]?internal_calls,900,1)
exten => 1337,4,GotoIf($["${digit}" = “3”]?internal_calls,777,1)
exten => 1337,5,Goto(1337,1)
And
[internal_calls]
include => IVR
(Ö)
Step 2.5 – Call forward
For each IPBX, on the /etc/asterisk/features.conf
[general]
parkext=>700
parkpos=>701-710
[featuremap]
blindxfer => # ;transfert inconditionnel
atxfer => * ;transfert conditionnel
For each IPBX in /etc/asterisk/extensions.conf
[internal_calls]
include => ParkedCalls
For each extensions on both IPBX in /etc/asterisk/extensions.conf, modify the extensions as well :
exten => NNN,2,Dial(SIP/NNN,20,T)
The third parameter ´ T ª will permet the calling part to forward the call.
ON REDEMARRE Asterisk : etc/init.d/asterisk restart
Step 2.6 – QoS configuration
/etc/asterisk/sip.conf
[general]
(Ö)
tos_sip=cs3
tos_audio=ef
/etc/asterisk/iax.conf
[general]
(Ö)
tos=ef
Step 2.7 – Logging
In /etc/asterisk/cdr.conf
[general]
enable=yes
[csv]
usegmtime=no
loguniqueid=yes
loguserfield=yes
Step 2.8 – Automation
In the /etc/asterisk/extensions.conf
exten => 777,1,Answer()
exten => 777,2,VoiceMailMain(${CALLERID(num)}@voicemail)
exten => 777,3,HungUp()
[macro-internal_calls]
exten => s,1,Answer()
exten => s,2,Dial(${ARG1},20,T)
exten => s,3,VoiceMail(${ARG1:4}@voicemail)
exten => s,4,PlayBack(vm-goodbye)
exten => s,5,Hangup()
[internal_calls]
exten => 500, 1, Macro(internal_calls,SIP/500)
exten => 501, 1, Macro(internal_calls,SIP/501)
OU
exten => _5XX, 1, Macro(internal_calls,SIP/${EXTEN})