2016-12-29 11:25:39 +01:00
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**Cette page a été importée automatiquement de notre ancien wiki mais n'a pas encore été révisée.**
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# Howto Asterisk
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<http://www.asterisk.org/community/documentation>
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## Installation
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~~~
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# aptitude install asterisk
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~~~
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On active son démarrage dans /etc/default/asterisk :
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~~~
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RUNASTERISK=yes
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~~~
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Puis on démarre :
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~~~
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# /etc/init.d/asterisk start
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Starting Asterisk PBX: asterisk.
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~~~
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Connexion en mode CLI :
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~~~
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# asterisk -r
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Asterisk 1.6.2.9-2+squeeze10, Copyright (C) 1999 - 2010 Digium, Inc. and others.
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Created by Mark Spencer <markster@digium.com>
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Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
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This is free software, with components licensed under the GNU General Public
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License version 2 and other licenses; you are welcome to redistribute it under
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certain conditions. Type 'core show license' for details.
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=========================================================================
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Connected to Asterisk 1.6.2.9-2+squeeze10 currently running on localhost (pid = 32537)
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*CLI>
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~~~
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Plus d'infos sur les commandes sur <http://www.voip-info.org/wiki/view/Asterisk+CLI>
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## Configuration
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La configuration se passe dans le répertoire /etc/asterisk. Pour un mode VoIP simple, les fichiers de configuration principaux sont
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* sip.conf
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* extensions.conf
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## Réseau
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Le protocole SIP utilise principalement le port UDP/5060 + des ports UDP divers...
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## Mode CLI
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Recharger la configuration Asterisk :
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~~~
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*CLI> reload
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~~~
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### SIP
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Lister les connexions SIP entrantes/sortantes :
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~~~
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*CLI> sip show peers
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Name/username Host Dyn Nat ACL Port Status
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[...]
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~~~
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Détails sur une connexion SIP particulière :
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~~~
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*CLI> sip show peer <PEER>
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~~~
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Lister les channels :
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~~~
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*CLI> sip show channels
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Peer User/ANR Call ID Format Hold Last Message Expiry
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1.2.3.4 (None) 53fa8add1dd3cba 0x0 (nothing) No Rx: OPTIONS
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192.168.0.21 (None) 32e1708b-5b55-4 0x0 (nothing) No Rx: REGISTER
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2 active SIP dialogs
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~~~
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Mode debug on / off pour SIP :
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~~~
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*CLI> sip set debug on
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SIP Debugging enabled
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*CLI> sip set debug off
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SIP Debugging Disabled
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~~~
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### Plan de numérotation
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Voir le plan de numérotation dans un contexte :
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~~~
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*CLI> dialplan show default
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[ Context 'default' created by 'pbx_config' ]
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'100' => 1. Answer() [pbx_config]
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2. Playback(demo-echotest) [pbx_config]
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3. Echo() [pbx_config]
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4. Playback(demo-echodone) [pbx_config]
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5. Hangup() [pbx_config]
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[...]
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~~~
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Recharger le plan de numérotation :
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~~~
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*CLI> dialplan reload
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Dialplan reloaded.
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~~~
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## Meeting Room
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2017-03-14 23:39:41 +01:00
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### Debian 7
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2016-12-29 11:25:39 +01:00
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Pour avoir des meeting rooms, il est nécessaire d'avoir le module DAHDI
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~~~
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# aptitude install linux-headers-`uname -r` dahdi-source asterisk-dahdi
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# m-a a-i dahdi
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# modprobe dahdi_dummy
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# chown -R asterisk: /dev/dahdi/
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~~~
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On peut ainsi configurer via meetme.conf :
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~~~
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conf => 1
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conf => 2
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conf => 3,1234
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~~~
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Et ensuite dans le plan de numérotation :
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~~~
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exten => 7001,1,MeetMe(1)
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exten => 7002,1,MeetMe(2)
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exten => 7003,1,MeetMe(3)
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~~~
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On peut lister les meeting rooms :
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~~~
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Meetme list
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Conf Num Parties Marked Activity Creation Locked
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2 0002 N/A 00:11:55 Static No
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1 0001 N/A 00:53:33 Static No
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* Total number of MeetMe users: 3
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~~~
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On peut lister les participants à une meeting room ...et même killer (pratique en cas de bug d'un utilisateur non déconnecté) :
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~~~
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Meetme list 1
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Meetme kick 1 8
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~~~
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2017-03-14 23:39:41 +01:00
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### Debian 8
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~~~
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exten => 7001,1,ConfBridge(1)
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exten => 7002,1,ConfBridge(2)
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exten => 7003,1,ConfBridge(3)
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~~~
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~~~
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CLI> confbridge list
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Conference Bridge Name Users Marked Locked?
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================================ ====== ====== ========
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2017-05-29 11:59:09 +02:00
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1 1 0 unlocked
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~~~
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~~~
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CLI> confbridge list 1
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Channel User Profile Bridge Profile Menu CallerID Muted
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============================== ================ ================ ================ ================ =====
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SIP/foobar-00001234 default_user default_bridge foobar No
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CLI> confbridge kick 1 SIP/foobar-00001234
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Participant 'SIP/foobar-00001234' kicked out of conference '1'
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2017-03-14 23:39:41 +01:00
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~~~
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2020-03-18 23:18:00 +01:00
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### Music On Hold (MOH)
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<https://www.voip-info.org/using-ffmpeg-to-convert-music-on-hold-files/>
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2017-03-14 23:39:41 +01:00
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2016-12-29 11:25:39 +01:00
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## Configuration téléphones
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Pour les téléphones Linksys/CISCO (SPA922, SPA901, SPA512G, SPA3102, PAP2T, etc.).
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On crée un entrée dans sip.conf :
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~~~
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[telephone1]
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type=friend
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host=dynamic
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username=LOGIN
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secret=PASS
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quality=yes
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nat=no
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canreinvite=no
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dtfmode=rfc2833
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allow=ulaw
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~~~
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Puis on configure le téléphone ainsi :
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~~~
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EXT1
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Proxy : IP
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Register : yes
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Display Name : telephone1
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User ID : LOGIN
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Password : PASS
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use Auth ID : no
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Auth ID : LOGIN
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~~~
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### Magic code Cisco / Linksys
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Composer `****` puis :
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~~~
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110# : IP address
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73738# puis 1 : reset
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7932# puis 1 : enable web on WAN
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2020-03-18 23:18:00 +01:00
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~~~
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