manuel du CLI tappez : help dans la CLI
et pour sortir comme 99% des applis lancées en shell avec ctrl+ C
manuel du CLI tappez : help dans la CLI
et pour sortir comme 99% des applis lancées en shell avec ctrl+ C
Bonsoir,
Code HTML:root@debian:~# asterisk -rvvvvv Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.8.11.0 currently running on debian (pid = 2080) Verbosity is at least 5 debian*CLI>
C'est ok la, si je veut faire des tests avec x-lite, quel sont les paramètres de Asterisk a indiquer à x-lite pour l'utiliser ?Code HTML:debian*CLI>sip show registry Host dnsmgr Username Refresh State Reg.Time freephonie.net:5060 N 0950xxxxxx 1785 Registered Tue, 1 SIP registrations. [Apr 10 15:25:53] NOTICE[2101]: chan_sip.c:26166 sip_poke_noanswer: Peer 'trunk-free' is now UNREACHABLE! Last qualify: 37 [Apr 10 15:26:03] NOTICE[2101]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (38ms / 2000ms) > doing dnsmgr_lookup for 'freephonie.net' [Apr 10 15:41:34] WARNING[2101]: chan_sip.c:20651 handle_response_register: Got 423 Interval too brief for service 0950xxxxxx@freephonie.net, minimum is 1800 seconds > doing dnsmgr_lookup for 'freephonie.net' > doing dnsmgr_lookup for 'freephonie.net' [Apr 10 15:44:06] NOTICE[2101]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Lagged. (3045ms / 2000ms) [Apr 10 15:44:17] NOTICE[2101]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (38ms / 2000ms) debian*CLI>
Dans "sip.conf" il ne faut pas indiquer le DNS de la freebox (192.168.0.254) quelque part ?
Merci.
Dernière modification par xunil2003 ; 10/04/2012 à 17h36.
Pour le compte free il faut mettre defaultexpiry=1800[Apr 10 15:41:34] WARNING[2101]: chan_sip.c:20651 handle_response_register: Got 423 Interval too brief for service 0950xxxxxx@freephonie.net, minimum is 1800 seconds
bonjour,
Merci c'est modifier
Mais je ne comprend pas ce que je doit indiquer a X-lite pour communiquer avec asterisk ?Code HTML:root@debian:~# asterisk -rvvvvv Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.8.11.0 currently running on debian (pid = 2327) Verbosity was 0 and is now 5 [Apr 10 17:43:43] NOTICE[2348]: chan_sip.c:26166 sip_poke_noanswer: Peer 'trunk-free' is now UNREACHABLE! Last qualify: 38 [Apr 10 17:43:53] NOTICE[2348]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (38ms / 2000ms) > doing dnsmgr_lookup for 'freephonie.net' > doing dnsmgr_lookup for 'freephonie.net' [Apr 10 18:17:59] NOTICE[2348]: chan_sip.c:26166 sip_poke_noanswer: Peer 'trunk-free' is now UNREACHABLE! Last qualify: 76 [Apr 10 18:18:09] NOTICE[2348]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (39ms / 2000ms) [Apr 10 18:20:13] NOTICE[2348]: chan_sip.c:26166 sip_poke_noanswer: Peer 'trunk-free' is now UNREACHABLE! Last qualify: 38 [Apr 10 18:20:23] NOTICE[2348]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (38ms / 2000ms) [Apr 10 18:23:27] NOTICE[2348]: chan_sip.c:26166 sip_poke_noanswer: Peer 'trunk-free' is now UNREACHABLE! Last qualify: 39 [Apr 10 18:23:37] NOTICE[2348]: chan_sip.c:20788 handle_response_peerpoke: Peer 'trunk-free' is now Reachable. (45ms / 2000ms) > doing dnsmgr_lookup for 'freephonie.net' > doing dnsmgr_lookup for 'freephonie.net' debian*CLI>
Merci.
Dernière modification par xunil2003 ; 10/04/2012 à 21h01.
Bonjour,
Je n'arrive pas a me faire identifier par Ekiga.
Dans sip.conf j'ai mis ceci
Avec Ekiga (ajouter un compte)Code HTML:[11];salle informatique type=friend username=poste11 secret=11 context=rdc quality=yes nat=yes canreinvite=no ;auth=md5 host=dynamic dtfmode=auto allow=ulaw mailbox=11 pickupgroup=1
Ekiga me répond " Impossible de s'inscrire"Code HTML:Nom : Salle informatique Registrar : 192.168.0.1 Utilisateur : poste11 Identifiant d'authentification : poste11 Mot de passe : 11 Delai : 360
Code HTML:root@debian:~# asterisk -rvvvvv Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.8.11.0 currently running on debian (pid = 2657) Verbosity was 0 and is now 5 debian*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 11/poste11 192.168.0.2 D N 5060 OK (31 ms) 12/poste12 (Unspecified) D N 0 UNKNOWN 13/chambre invité (Unspecified) D N 0 UNKNOWN 14/cuisine d'été (Unspecified) D N 0 UNKNOWN freephonie_in 212.27.52.5 N 5060 OK (37 ms) freephonie_out/0950140909 212.27.52.5 N 5060 OK (37 ms) 6 sip peers [Monitored: 3 online, 3 offline Unmonitored: 0 online, 0 offline] [Apr 11 11:28:28] WARNING[2678]: chan_sip.c:14399 check_auth: username mismatch, have <11>, digest has <poste11> [Apr 11 11:28:28] NOTICE[2678]: chan_sip.c:24929 handle_request_register: Registration from '<sip:11@192.168.0.1>' failed for '192.168.0.2:5060' - Username/auth name mismatch [Apr 11 11:28:44] NOTICE[2678]: chan_sip.c:24929 handle_request_register: Registration from '<sip:poste11@192.168.0.1>' failed for '192.168.0.2:5060' - No matching peer found [Apr 11 11:28:44] NOTICE[2678]: chan_sip.c:24929 handle_request_register: Registration from '<sip:poste11@192.168.0.1>' failed for '192.168.0.2:5060' - No matching peer found debian*CLI>
Ou est mon erreur , je rempli mal les champs d'Ekiga ?
Merci
Dernière modification par xunil2003 ; 11/04/2012 à 15h49.
Bonjour,
C'est bon ,'j'ai trouvé, j'ai réussi a ajouter mon compte dans Ekiga, mais j'ai un problème avec la messagerie d'asterisk.
Il me répond "Inscrit" .Code HTML:Nom : poste11 Registrar : 192.168.0.1 Utilisateur : 11 Identifiant d'authentification : 11 Mot de passe : 11 Delai : 360
Messagerie
Cependant pour la messagerie quand je compose le 700 j'ai ma messagerie.
(asterisk) Messagerie asterisk, veuillez composer votre numéro de boite vocale
(moi) Je tape : 11
(asterisk) Mot de passe
(moi) Je tape 11
(asterisk)Access refusé, veuillez recomposer votre numéro de boite vocale
Code HTML:root@debian:~# asterisk -rvvvvv Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.8.11.0 currently running on debian (pid = 2889) Verbosity was 0 and is now 5 debian*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 11/poste11 192.168.0.2 D N 5060 OK (31 ms) 12/poste12 (Unspecified) D N 0 UNKNOWN 13/chambre invité (Unspecified) D N 0 UNKNOWN 14/cuisine d'été (Unspecified) D N 0 UNKNOWN freephonie_in 212.27.52.5 N 5060 OK (37 ms) freephonie_out/0950140909 212.27.52.5 N 5060 OK (37 ms) 6 sip peers [Monitored: 3 online, 3 offline Unmonitored: 0 online, 0 offline] == Using SIP RTP CoS mark 5 -- Executing [12@rdc:1] Dial("SIP/11-00000000", ", 30, wW") in new stack [Apr 11 12:31:40] WARNING[2937]: app_dial.c:1949 dial_exec_full: Dial requires an argument (technology/number) == Spawn extension (rdc, 12, 1) exited non-zero on 'SIP/11-00000000' == Using SIP RTP CoS mark 5 -- Executing [700@rdc:1] VoiceMailMain("SIP/11-00000001", "") in new stack -- <SIP/11-00000001> Playing 'vm-login.slin' (language 'fr') -- <SIP/11-00000001> Playing 'vm-password.slin' (language 'fr') -- Incorrect password '' for user '11' (context = default) -- <SIP/11-00000001> Playing 'vm-incorrect-mailbox.slin' (language 'fr') -- <SIP/11-00000001> Playing 'vm-password.slin' (language 'fr') -- Incorrect password '' for user '11' (context = default) -- <SIP/11-00000001> Playing 'vm-incorrect-mailbox.slin' (language 'fr') -- <SIP/11-00000001> Playing 'vm-password.slin' (language 'fr') -- Incorrect password '' for user '11' (context = default) -- <SIP/11-00000001> Playing 'vm-incorrect.slin' (language 'fr') -- <SIP/11-00000001> Playing 'vm-goodbye.slin' (language 'fr') -- Auto fallthrough, channel 'SIP/11-00000001' status is 'UNKNOWN' == Using SIP RTP CoS mark 5 -- Executing [700@rdc:1] VoiceMailMain("SIP/11-00000002", "") in new stack -- <SIP/11-00000002> Playing 'vm-login.slin' (language 'fr') -- <SIP/11-00000002> Playing 'vm-password.slin' (language 'fr') -- Incorrect password '70' for user '11' (context = default) -- <SIP/11-00000002> Playing 'vm-incorrect-mailbox.slin' (language 'fr') [Apr 11 12:45:10] WARNING[2950]: app_voicemail.c:9759 vm_authenticate: Couldn't read username == Using SIP RTP CoS mark 5 -- Executing [700@rdc:1] VoiceMailMain("SIP/11-00000003", "") in new stack -- <SIP/11-00000003> Playing 'vm-login.slin' (language 'fr') -- <SIP/11-00000003> Playing 'vm-password.slin' (language 'fr') -- Incorrect password '11' for user '11' (context = default) -- <SIP/11-00000003> Playing 'vm-incorrect-mailbox.slin' (language 'fr') [Apr 11 12:45:35] WARNING[2951]: app_voicemail.c:9759 vm_authenticate: Couldn't read username > doing dnsmgr_lookup for 'freephonie.net' > doing dnsmgr_lookup for 'freephonie.net' debian*CLI>
Je met le fichier sip.conf et extentions.conf dans la 2em partie.
Ou est l'erreur , que ce passe t'il ?
Merci.
Dernière modification par xunil2003 ; 11/04/2012 à 15h56.
2em partie
sip.conf
extentions.confCode HTML:[general] language=fr bindport=5060 bindaddr=0.0.0.0 context=default srvlookup=no externip = 78.xxx.xx.xxx (Mon Ip internet) localnet = 192.168.0.0/255.255.255.0 ;localnet=192.168.5.0/255.255.255.0 defaultexpirey=1800 dtmfmode=auto relaxdtmf=yes qualify=yes register= 09xxxxxx:motdepasse@freephonie.net disallow=all allow=ulaw allow=alaw allow=gsm [freephonie_out] nat=yes type=peer disallow=all allow=alaw allow=ulaw host=freephonie.net secret=xxxxxx(mon mot de passe) fromuser=09xxxxxx username=09xxxxxxx dtmfmode=auto qualify=yes fromdomain=freephonie.net context=default [freephonie_in] type=peer context=maison ;context=fromfree host=freephonie.net qualify=yes allow=all dtmfmode=auto ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;; ;Definition des comptes de telephone ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;; [11];salle informatique type=friend username=poste11 secret=11 context=rdc quality=yes nat=yes canreinvite=no ;auth=md5 host=dynamic dtfmode=auto allow=ulaw mailbox=11 pickupgroup=1 [12];Salle de soprt type=friend username=poste12 secret=12 context=rdc quality=yes nat=yes canreinvite=no ;auth=md5 host=dynamic dtfmode=auto allow=ulaw mailbox=12 pickupgroup=1 [13];chambre invité type=friend username=chambre invité secret=chambre invité context=rdc quality=yes nat=yes canreinvite=no ;auth=md5 host=dynamic dtfmode=auto allow=ulaw mailbox=13 pickupgroup=1 [14];cuisine d'été type=friend username=cuisine d'été secret=cuisine d'été context=rdc quality=yes nat=yes canreinvite=no ;auth=md5 host=dynamic dtfmode=auto allow=ulaw mailbox=14 pickupgroup=1
Merci.Code HTML:[general] ; option de protection du dialplan static=yes ; dialplan statique writeprotect=yes ; on ne peut modifier le dialplan via le CLI d'asterisk clearglobals=yes ; on recalcule les variables globales a chaque redemarrage d'asterisk exten => _11.,1,Pickup(${EXTEN:3}) [globals] ;variables globales (ne pas modifier) DYNAMIC_FEATURES => automon SALLEINFORMATIQUE=SIP/11 SALLEDESPORT=SIP/12 CHAMBREDEINVITE=SIP/13 CUISINEDETE=SIP/14 LEASEINFO=SIP/15 MADEFORDANCE=SIP/16 [maison] ;[fromfree] ; on declare le contexte de reception d'appels depuis freephonie (redirection vers le menu interactif) exten => s,1,Goto(accueil,666,1) ; Horloge parlante exten = 102,1,Answer exten = 102,2,SayUnixTime(,CET,kM) exten = 102,3,Hangup [default] ;section des parametres par defaut. include => parkedcalls ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;; ;Configuration du menu interactif ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;; [accueil] ; définition d’un contexte pour l’accueil exten => s,1,Answer() exten => s,n,Background(${sounds_path}accueil) exten => s,n,WaitExten(10) ;on definit la redirection vers le bon contexte exten => 1,1,Goto(cimedia,777,1) exten => 2,1,Goto(leaseinfo,888,1) exten => 3,1,Goto(madefordance,999,1) ;En cas de mauvaise saisie exten => i,1,Playback(${sounds_path}agent-incorrect) exten => i,n,Goto(accueil,666,1) ;En cas de timeout exten => t,1,Playback(${sounds_path}vm-goodbye) exten => t,n,Hangup() [cimedia] ; menu interactif cimedia exten => 777,1,Background(${sounds_path}menu_ci) exten => 777,n,WaitExten(10) ;Si l'appelant à séelectionne un bon choix ds le menu on fait sonner le softphone correspondant exten => 1,1,Dial(${DIRECTION}, 30, wW) exten => 1,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 1,n(unavail),Voicemail(10,u) exten => 1,n,Hangup() exten => 1,n(busy),VoiceMail(10,b) exten => 1,n,Hangup() exten => 2,1,Dial(${COMMERCIAL}, 30, wW) exten => 2,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 2,n(unavail),Voicemail(11,u) exten => 2,n,Hangup() exten => 2,n(busy),VoiceMail(11,b) exten => 2,n,Hangup() exten => 3,1,Dial(${SAV}, 30, wW) exten => 3,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 3,n(unavail),Voicemail(12,u) exten => 3,n,Hangup() exten => 3,n(busy),VoiceMail(12,b) exten => 3,n,Hangup() exten => 4,1,Dial(${TECHNIQUE}, 30, wW) exten => 4,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 4,n(unavail),Voicemail(13,u) exten => 4,n,Hangup() exten => 4,n(busy),VoiceMail(13,b) exten => 4,n,Hangup() ;Si mauvaise saisie exten => i,1,Playback(${sounds_path}agent-incorrect) exten => i,n,Goto(cimedia,777,1) ;Si Timeout exten => t,1,Playback(${sounds_path}vm-goodbye) exten => t,n,Hangup() [leaseinfo] ; menu interactif leaseinfo ;exten => 888,1,Background(${sounds_path}menu_ci) ;exten => 888,n,WaitExten(10) ;Si l'appelant à séelectionne un bon choix ds le menu on fait sonner le softphone correspondant exten => 888,1,Dial(${DIRECTION}, 30, wW) exten => 888,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 888,n(unavail),Voicemail(10,u) exten => 888,n,Hangup() exten => 888,n(busy),VoiceMail(10,b) exten => 888,n,Hangup() ;Si mauvaise saisie exten => i,1,Playback(${sounds_path}agent-incorrect) exten => i,n,Goto(leaseinfo,888,1) ;Si Timeout exten => t,1,Playback(${sounds_path}vm-goodbye) exten => t,n,Hangup() [madefordance] ; menu interactif ent3 ;exten => 999,1,Background(${sounds_path}menu_ci) ;exten => 999,n,WaitExten(10) ;Si l'appelant à séelectionne un bon choix ds le menu on fait sonner le softphone correspondant exten => 999,1,Dial(${MADEFORDANCE}, 30, wW) exten => 999,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 999,n(unavail),Voicemail(15,u) exten => 999,n,Hangup() exten => 999,n(busy),VoiceMail(15,b) exten => 999,n,Hangup() ;Si mauvaise saisie exten => i,1,Playback(${sounds_path}agent-incorrect) exten => i,n,Goto(madefordance,999,1) ;Si Timeout exten => t,1,Playback(${sounds_path}vm-goodbye) exten => t,n,Hangup() ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;; ;Configuration des comptes locaux ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;; [rdc] ; on declare le contexte local qu'on a specifie dans le sip.conf ; numeros "locaux" exten => 11,1,Dial(${COMMERCIAL}, 30, wW) ; quand on compose le 11, le softphone branché sur le lien "cCOMMERCIAL" sonnera exten => 11,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 11,n(unavail),Voicemail(11,u) exten => 11,n,Hangup() exten => 11,n(busy),VoiceMail(11,b) exten => 11,n,Hangup() exten => 12,1,Dial(${SAV}, 30, wW) ; quand on compose le 12, le softphone branché sur le lien "cSAV" sonnera exten => 12,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 12,n(unavail),Voicemail(12,u) exten => 12,n,Hangup() exten => 12,n(busy),VoiceMail(12,b) exten => 12,n,Hangup() exten => 13,1,Dial(${TECHNIQUE}, 30, wW) ; quand on compose le 13, le softphone branché sur le lien "pTECHNIQUE" sonnera exten => 13,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 13,n(unavail),Voicemail(13,u) exten => 13,n,Hangup() exten => 13,n(busy),VoiceMail(13,b) exten => 13,n,Hangup() exten => 14,1,Dial(${COMPTA}, 30, wW) ; quand on compose le 14, le softphone branché sur le lien "pCOMPTA" sonnera exten => 14,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 14,n(unavail),Voicemail(14,u) exten => 14,n,Hangup() exten => 14,n(busy),VoiceMail(14,b) exten => 14,n,Hangup() exten => 15,1,Dial(${LEASEINFO}, 30, wW) ; quand on compose le 15, le softphone branché sur le lien ""LEASEINFO" sonnera exten => 15,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 15,n(unavail),Voicemail(15,u) exten => 15,n,Hangup() exten => 15,n(busy),VoiceMail(15,b) exten => 15,n,Hangup() exten => 16,1,Dial(${MADEFORDANCE}, 30, wW) ; quand on compose le 16, le softphone branché sur le lien "pMADEFORDANCE" sonnera exten => 16,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) exten => 16,n(unavail),Voicemail(16,u) exten => 16,n,Hangup() exten => 16,n(busy),VoiceMail(16,b) exten => 16,n,Hangup() ;Extension pour appeler directement le repondeur exten => 700,1,VoicemailMain() ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;; ;Configuration des appels sortants ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;;;;;;;;;;;;;;; ; numeros externes ; quand on compose un numero qui commence par 0,on utilise le lien "freephonie" ;et on passe le numero au peer en otant le premier digit. exten => _0.,1,Dial(SIP/freephonie_out/${EXTEN})