La Fibre
Datacenter et équipements réseaux => Routeurs => 
Remplacer la LiveBox par un routeur => Discussion démarrée par: jeremy64 le 18 avril 2017 à 21:44:44
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Bonsoir,je souhaitais savoir si quelqu'un avait reussi à faire fonctionner Xivo pour recuperer la telephonie d'orange. J'ai remplacé la livebox par une distri pfsense.
J'ai sur le meme serveur un VM Pfsense qui recupere donc internet. ( la tout est ok).
Sur une autre VM j'ai debian avec Siproxd + Xivo. Le siproxd est fonctionnel puisque aujourd hui mon tel IP C530IP se connecte sur Siproxd pour pouvoir appeler et recevoir mais je souhaite pouvoir gerer un pré-dec / musique d'attente / horaire...
Pour Xivo, quand j'essai d'appelé j'ai un message " il n'y a pas d'usager correspondant au numero demandé".
Le CLI nous donne :
[Apr 18 21:25:37] == Using SIP RTP CoS mark 5
[Apr 18 21:25:37] -- Executing [s@default:1] NoOp("SIP/Entrant_livebox-0000000c", "") in new stack
[Apr 18 21:25:37] -- Executing [s@default:2] GotoIf("SIP/Entrant_livebox-0000000c", "1?:not-sip") in new stack
[Apr 18 21:25:37] -- Executing [s@default:3] GotoIf("SIP/Entrant_livebox-0000000c", "1?:error-loop") in new stack
[Apr 18 21:25:37] -- Executing [s@default:4] Set("SIP/Entrant_livebox-0000000c", "XIVO_DID_NEXT_EXTEN=+33547*******") in new stack
[Apr 18 21:25:37] -- Executing [s@default:5] Set("SIP/Entrant_livebox-0000000c", "XIVO_FROM_S=1") in new stack
[Apr 18 21:25:37] -- Executing [s@default:6] Goto("SIP/Entrant_livebox-0000000c", "from-extern,+33547*******,1") in new stack
[Apr 18 21:25:37] -- Goto (from-extern,+33547*****,1)
[Apr 18 21:25:37] -- Executing [+33547******@from-extern:1] Goto("SIP/Entrant_livebox-0000000c", "33547******,1") in new stack
[Apr 18 21:25:37] -- Goto (from-extern,33547*****,1)
[Apr 18 21:25:37] -- Channel 'SIP/Entrant_livebox-0000000c' sent to invalid extension: context,exten,priority=from-extern,33547*****,1
[Apr 18 21:25:37] -- Executing [i@from-extern:1] Playback("SIP/Entrant_livebox-0000000c", "no-user-find") in new stack
[Apr 18 21:25:37] == Using SIP RTP CoS mark 5
[Apr 18 21:25:37] -- Executing [s@default:1] NoOp("SIP/Entrant_livebox-0000000d", "") in new stack
[Apr 18 21:25:37] -- Executing [s@default:2] GotoIf("SIP/Entrant_livebox-0000000d", "1?:not-sip") in new stack
[Apr 18 21:25:37] -- Executing [s@default:3] GotoIf("SIP/Entrant_livebox-0000000d", "1?:error-loop") in new stack
[Apr 18 21:25:37] -- Executing [s@default:4] Set("SIP/Entrant_livebox-0000000d", "XIVO_DID_NEXT_EXTEN=+33547*****") in new stack
[Apr 18 21:25:37] -- Executing [s@default:5] Set("SIP/Entrant_livebox-0000000d", "XIVO_FROM_S=1") in new stack
[Apr 18 21:25:37] -- Executing [s@default:6] Goto("SIP/Entrant_livebox-0000000d", "from-extern,+33547******,1") in new stack
[Apr 18 21:25:37] -- Goto (from-extern,+33547*****,1)
[Apr 18 21:25:37] -- Executing [+33547******@from-extern:1] Goto("SIP/Entrant_livebox-0000000d", "33547*****,1") in new stack
[Apr 18 21:25:37] -- Goto (from-extern,33547******,1)
[Apr 18 21:25:37] -- Channel 'SIP/Entrant_livebox-0000000d' sent to invalid extension: context,exten,priority=from-extern,33547******,1
[Apr 18 21:25:37] -- Executing [i@from-extern:1] Playback("SIP/Entrant_livebox-0000000d", "no-user-find") in new stack
[Apr 18 21:25:37] == Using SIP RTP CoS mark 5
[Apr 18 21:25:37] -- Executing [s@default:1] NoOp("SIP/Entrant_livebox-0000000e", "") in new stack
[Apr 18 21:25:37] -- Executing [s@default:2] GotoIf("SIP/Entrant_livebox-0000000e", "1?:not-sip") in new stack
[Apr 18 21:25:37] -- Executing [s@default:3] GotoIf("SIP/Entrant_livebox-0000000e", "1?:error-loop") in new stack
[Apr 18 21:25:37] -- Executing [s@default:4] Set("SIP/Entrant_livebox-0000000e", "XIVO_DID_NEXT_EXTEN=+33547******") in new stack
[Apr 18 21:25:37] -- Executing [s@default:5] Set("SIP/Entrant_livebox-0000000e", "XIVO_FROM_S=1") in new stack
[Apr 18 21:25:37] -- Executing [s@default:6] Goto("SIP/Entrant_livebox-0000000e", "from-extern,+33547******,1") in new stack
[Apr 18 21:25:37] -- Goto (from-extern,+33547******,1)
[Apr 18 21:25:37] -- Executing [+33547******@from-extern:1] Goto("SIP/Entrant_livebox-0000000e", "33547******from-extern,33547*****,1
[Apr 18 21:25:37] -- Executing [i@from-extern:1] Playback("SIP/Entrant_livebox-0000000e", "no-user-find") in new stack
[Apr 18 21:25:37] -- <SIP/Entrant_livebox-0000000c> Playing 'no-user-find.slin' (language 'fr_FR')
[Apr 18 21:25:37] -- <SIP/Entrant_livebox-0000000d> Playing 'no-user-find.slin' (language 'fr_FR')
[Apr 18 21:25:37] > 0x7f30f4005970 -- Probation passed - setting RTP source address to 192.168.1.140:7076
[Apr 18 21:25:37] -- <SIP/Entrant_livebox-0000000e> Playing 'no-user-find.slin' (language 'fr_FR')
[Apr 18 21:25:41] -- Executing [i@from-extern:2] Hangup("SIP/Entrant_livebox-0000000c", "") in new stack
[Apr 18 21:25:41] == Spawn extension (from-extern, i, 2) exited non-zero on 'SIP/Entrant_livebox-0000000c'
[Apr 18 21:25:41] -- Executing [i@from-extern:2] Hangup("SIP/Entrant_livebox-0000000d", "") in new stack
[Apr 18 21:25:41] == Spawn extension (from-extern, i, 2) exited non-zero on 'SIP/Entrant_livebox-0000000d'
[Apr 18 21:25:41] -- Executing [i@from-extern:2] Hangup("SIP/Entrant_livebox-0000000e", "") in new stack
[Apr 18 21:25:41] == Spawn extension (from-extern, i, 2) exited non-zero on 'SIP/Entrant_livebox-0000000e'
puis
WARNING[7719]: chan_sip.c:4064 retrans_pkt: Retransmission timeout reached on transmission SD05kbe01-6c24aff90c4a5e3b13d53fc177b65762-1eurdu1 for seqno 21100153 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Pour les appels sortant c'est pareil.
Si quelqu'un a une idée sur ce type d'erreur, je suis preneur.
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Salut,
Tu n'aurais pas oublié d'associer un poste à la DID/SDA ?
Les appels internes fonctionnent correctement ?