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Voir la version complète : Interconnexion IPBX<--->OXE alcatel R10



bwen
28/09/2016, 12h47
Bonjour,

Je tente de mettre en place un trunk SIP entre un OXE alcatel et un asterisk (xivo) mais je ne parviens pas a m'en sortir.
J'ai configuré les faisceaux, passerelle SIP , table de routage ARS etc selon des exemples que j'ai pu trouver sur les forums, le trunk semble bien etabli mais aucun n'appel n'abouti, que ce soit dans un sens ou dans l'autre. Je me concentre d'abord sur le fait de faire fonctionner les appels de l'IPBX vers l'OXE, quelque soit le numero composé (interne ou externe) je tombe inlassablement sur un 404 not found, je sais plus ou chercher, si quelqu'un pouvait me donner quelques pistes a explorer, je lui en serait gré ;-)
Merci


[Sep 27 13:58:20] == Using SIP RTP CoS mark 5
[Sep 27 13:58:20] -- Executing [2266@default:1] Set("SIP/ifvdbq1h-00000004", "XIVO_BASE_CONTEXT=default") in new stack
[Sep 27 13:58:20] -- Executing [2266@default:2] Set("SIP/ifvdbq1h-00000004", "XIVO_BASE_EXTEN=2266") in new stack
[Sep 27 13:58:20] -- Executing [2266@default:3] Gosub("SIP/ifvdbq1h-00000004", "outcall,s,1(3,)") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:1] Set("SIP/ifvdbq1h-00000004", "XIVO_DSTID=3") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:2] Set("SIP/ifvdbq1h-00000004", "XIVO_PRESUBR_GLOBAL_NAME=OUTCALL") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:3] Set("SIP/ifvdbq1h-00000004", "XIVO_SRCNUM=667") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:4] Set("SIP/ifvdbq1h-00000004", "XIVO_DSTNUM=2266") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:5] Set("SIP/ifvdbq1h-00000004", "XIVO_CONTEXT=default") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:6] Gosub("SIP/ifvdbq1h-00000004", "originate-caller-id,s,1") in new stack
[Sep 27 13:58:20] -- Executing [s@originate-caller-id:1] GotoIf("SIP/ifvdbq1h-00000004", "0?:name") in new stack
[Sep 27 13:58:20] -- Goto (originate-caller-id,s,3)
[Sep 27 13:58:20] -- Executing [s@originate-caller-id:3] GotoIf("SIP/ifvdbq1h-00000004", "0?:fix") in new stack
[Sep 27 13:58:20] -- Goto (originate-caller-id,s,5)
[Sep 27 13:58:20] -- Executing [s@originate-caller-id:5] GotoIf("SIP/ifvdbq1h-00000004", "?:end") in new stack
[Sep 27 13:58:20] -- Goto (originate-caller-id,s,8)
[Sep 27 13:58:20] -- Executing [s@originate-caller-id:8] Return("SIP/ifvdbq1h-00000004", "") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:7] AGI("SIP/ifvdbq1h-00000004", "agi://127.0.0.1/outgoing_user_set_features") in new stack
[Sep 27 13:58:20] agi://127.0.0.1/outgoing_user_set_features: AGI handler 'outgoing_user_set_features' successfully executed
[Sep 27 13:58:20] -- <SIP/ifvdbq1h-00000004>AGI Script agi://127.0.0.1/outgoing_user_set_features completed, returning 0
[Sep 27 13:58:20] -- Executing [s@outcall:8] Gosub("SIP/ifvdbq1h-00000004", "xivo-subroutine,s,1()") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-subroutine:1] GotoIf("SIP/ifvdbq1h-00000004", "?:nosubroutine") in new stack
[Sep 27 13:58:20] -- Goto (xivo-subroutine,s,4)
[Sep 27 13:58:20] -- Executing [s@xivo-subroutine:4] Return("SIP/ifvdbq1h-00000004", "") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:9] Gosub("SIP/ifvdbq1h-00000004", "xivo-user_rights_check,s,1") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-user_rights_check:1] AGI("SIP/ifvdbq1h-00000004", "agi://127.0.0.1/user_set_call_rights") in new stack
[Sep 27 13:58:20] agi://127.0.0.1/user_set_call_rights: AGI handler 'user_set_call_rights' successfully executed
[Sep 27 13:58:20] -- <SIP/ifvdbq1h-00000004>AGI Script agi://127.0.0.1/user_set_call_rights completed, returning 0
[Sep 27 13:58:20] -- Executing [s@xivo-user_rights_check:2] GotoIf("SIP/ifvdbq1h-00000004", "ALLOW?:error,1") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-user_rights_check:3] GotoIf("SIP/ifvdbq1h-00000004", "1?allow,1") in new stack
[Sep 27 13:58:20] -- Goto (xivo-user_rights_check,allow,1)
[Sep 27 13:58:20] -- Executing [allow@xivo-user_rights_check:1] NoOp("SIP/ifvdbq1h-00000004", "User allowed to make call") in new stack
[Sep 27 13:58:20] -- Executing [allow@xivo-user_rights_check:2] Return("SIP/ifvdbq1h-00000004", "") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:10] AGI("SIP/ifvdbq1h-00000004", "agi://127.0.0.1/check_schedule") in new stack
[Sep 27 13:58:20] agi://127.0.0.1/check_schedule: AGI handler 'check_schedule' successfully executed
[Sep 27 13:58:20] -- <SIP/ifvdbq1h-00000004>AGI Script agi://127.0.0.1/check_schedule completed, returning 0
[Sep 27 13:58:20] -- Executing [s@outcall:11] GotoIf("SIP/ifvdbq1h-00000004", "0?CLOSED,1") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:12] GotoIf("SIP/ifvdbq1h-00000004", "?:14") in new stack
[Sep 27 13:58:20] -- Goto (outcall,s,14)
[Sep 27 13:58:20] -- Executing [s@outcall:14] GotoIf("SIP/ifvdbq1h-00000004", "SIP/OXE?:error,1") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:15] Set("SIP/ifvdbq1h-00000004", "TRUNKINDEX=0") in new stack
[Sep 27 13:58:20] -- Executing [s@outcall:16] Goto("SIP/ifvdbq1h-00000004", "dial,1") in new stack
[Sep 27 13:58:20] -- Goto (outcall,dial,1)
[Sep 27 13:58:20] -- Executing [dial@outcall:1] Set("SIP/ifvdbq1h-00000004", "INTERFACE=SIP/OXE") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:2] Set("SIP/ifvdbq1h-00000004", "TRUNKEXTEN=2266") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:3] Set("SIP/ifvdbq1h-00000004", "TRUNKSUFFIX=") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:4] Gosub("SIP/ifvdbq1h-00000004", "xivo-global-subroutine,s,1") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-global-subroutine:1] GotoIf("SIP/ifvdbq1h-00000004", "1?:return") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-global-subroutine:2] GotoIf("SIP/ifvdbq1h-00000004", "OUTCALL?:return") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-global-subroutine:3] GotoIf("SIP/ifvdbq1h-00000004", "xivo-subrgbl-outcall?:return") in new stack
[Sep 27 13:58:20] -- Executing [s@xivo-global-subroutine:4] GotoIf("SIP/ifvdbq1h-00000004", "0?:return") in new stack
[Sep 27 13:58:20] -- Goto (xivo-global-subroutine,s,6)
[Sep 27 13:58:20] -- Executing [s@xivo-global-subroutine:6] Return("SIP/ifvdbq1h-00000004", "") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:5] CELGenUserEvent("SIP/ifvdbq1h-00000004", "XIVO_OUTCALL") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:6] Set("SIP/ifvdbq1h-00000004", "CONNECTEDLINE(num,i)=2266") in new stack
[Sep 27 13:58:20] -- Executing [dial@outcall:7] Dial("SIP/ifvdbq1h-00000004", "SIP/OXE/2266,,o(2266)") in new stack
[Sep 27 13:58:20] == Using SIP RTP CoS mark 5
[Sep 27 13:58:20] Audio is at 10642
[Sep 27 13:58:20] Adding codec alaw to SDP
[Sep 27 13:58:20] Adding codec ulaw to SDP
[Sep 27 13:58:20] Adding codec g723 to SDP
[Sep 27 13:58:20] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 27 13:58:20] Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
[Sep 27 13:58:20] INVITE sip:2266@XXX.XXX.XXX.XXX:5060 SIP/2.0
[Sep 27 13:58:20] Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK509a136f
[Sep 27 13:58:20] Max-Forwards: 70
[Sep 27 13:58:20] From: "jean-marc " <sip:667@XXX.XXX.XXX.XXX>;tag=as1944ec45
[Sep 27 13:58:20] To: <sip:2266@XXX.XXX.XXX.XXX:5060>
[Sep 27 13:58:20] Contact: <sip:667@XXX.XXX.XXX.XXX:5060>
[Sep 27 13:58:20] Call-ID: 50efa5d654782cd94aa9658e136beb0f@XXX.XXX.XXX.XXX
[Sep 27 13:58:20] CSeq: 102 INVITE
[Sep 27 13:58:20] User-Agent: XIVO PBX
[Sep 27 13:58:20] Date: Tue, 27 Sep 2016 11:58:20 GMT
[Sep 27 13:58:20] Session-Expires: 1800
[Sep 27 13:58:20] Min-SE: 300
[Sep 27 13:58:20] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 27 13:58:20] Supported: replaces, timer
[Sep 27 13:58:20] Remote-Party-ID: "jean-marc " <sip:667@XXX.XXX.XXX.XXX>;party=calling;privacy=off;screen=no
[Sep 27 13:58:20] Content-Type: application/sdp
[Sep 27 13:58:20] Content-Length: 317
[Sep 27 13:58:20]
[Sep 27 13:58:20] v=0
[Sep 27 13:58:20] o=root 804414646 804414646 IN IP4 XXX.XXX.XXX.XXX
[Sep 27 13:58:20] s=Asterisk PBX 13.9.1
[Sep 27 13:58:20] c=IN IP4 XXX.XXX.XXX.XXX
[Sep 27 13:58:20] t=0 0
[Sep 27 13:58:20] m=audio 10642 RTP/AVP 8 0 4 101
[Sep 27 13:58:20] a=rtpmap:8 PCMA/8000
[Sep 27 13:58:20] a=rtpmap:0 PCMU/8000
[Sep 27 13:58:20] a=rtpmap:4 G723/8000
[Sep 27 13:58:20] a=fmtp:4 annexa=no
[Sep 27 13:58:20] a=rtpmap:101 telephone-event/8000
[Sep 27 13:58:20] a=fmtp:101 0-16
[Sep 27 13:58:20] a=ptime:20
[Sep 27 13:58:20] a=maxptime:150
[Sep 27 13:58:20] a=sendrecv
[Sep 27 13:58:20]
[Sep 27 13:58:20] ---
[Sep 27 13:58:20] -- Called SIP/OXE/2266
[Sep 27 13:58:20]
[Sep 27 13:58:20] <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
[Sep 27 13:58:20] SIP/2.0 100 Trying
[Sep 27 13:58:20] To: <sip:2266@XXX.XXX.XXX.XXX:5060>
[Sep 27 13:58:20] From: "jean-marc " <sip:667@XXX.XXX.XXX.XXX>;tag=as1944ec45
[Sep 27 13:58:20] Call-ID: 50efa5d654782cd94aa9658e136beb0f@XXX.XXX.XXX.XXX
[Sep 27 13:58:20] CSeq: 102 INVITE
[Sep 27 13:58:20] Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK509a136f
[Sep 27 13:58:20] Content-Length: 0

bwen
28/09/2016, 12h47
la fin du sip debug..


[Sep 27 13:58:20]
[Sep 27 13:58:20]
[Sep 27 13:58:20] <------------->
[Sep 27 13:58:20] --- (7 headers 0 lines) ---
[Sep 27 13:58:20]
[Sep 27 13:58:20] <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
[Sep 27 13:58:20] SIP/2.0 404 Not Found
[Sep 27 13:58:20] Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
[Sep 27 13:58:20] User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
[Sep 27 13:58:20] To: <sip:2266@XXX.XXX.XXX.XXX:5060>;tag=8662dea65b7c4794168f018c95a5a01c
[Sep 27 13:58:20] From: "jean-marc " <sip:667@XXX.XXX.XXX.XXX>;tag=as1944ec45
[Sep 27 13:58:20] Call-ID: 50efa5d654782cd94aa9658e136beb0f@XXX.XXX.XXX.XXX
[Sep 27 13:58:20] CSeq: 102 INVITE
[Sep 27 13:58:20] Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK509a136f
[Sep 27 13:58:20] Content-Length: 0
[Sep 27 13:58:20]
[Sep 27 13:58:20]
[Sep 27 13:58:20] <------------->
[Sep 27 13:58:20] --- (9 headers 0 lines) ---
[Sep 27 13:58:20] Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
[Sep 27 13:58:20] ACK sip:2266@XXX.XXX.XXX.XXX:5060 SIP/2.0
[Sep 27 13:58:20] Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK509a136f
[Sep 27 13:58:20] Max-Forwards: 70
[Sep 27 13:58:20] From: "jean-marc " <sip:667@XXX.XXX.XXX.XXX>;tag=as1944ec45
[Sep 27 13:58:20] To: <sip:2266@XXX.XXX.XXX.XXX:5060>;tag=8662dea65b7c4794168f018c95a5a01c
[Sep 27 13:58:20] Contact: <sip:667@XXX.XXX.XXX.XXX:5060>
[Sep 27 13:58:20] Call-ID: 50efa5d654782cd94aa9658e136beb0f@XXX.XXX.XXX.XXX
[Sep 27 13:58:20] CSeq: 102 ACK
[Sep 27 13:58:20] User-Agent: XIVO PBX
[Sep 27 13:58:20] Content-Length: 0
[Sep 27 13:58:20]
[Sep 27 13:58:20]
[Sep 27 13:58:20] ---
[Sep 27 13:58:20] Scheduling destruction of SIP dialog '50efa5d654782cd94aa9658e136beb0f@XXX.XXX.XXX.XXX' in 32000 ms (Method: INVITE)

jean
28/09/2016, 20h31
la bonne nouvelle, c'est que ca cause en sip entre le oxo et xivo - et le oxo n'arrive pas à trouver le no 2266, et renvoie not found - pourquoi, ca, faut voir avec le oxo

bwen
29/09/2016, 10h14
la bonne nouvelle, c'est que ca cause en sip entre le oxo et xivo - et le oxo n'arrive pas à trouver le no 2266, et renvoie not found - pourquoi, ca, faut voir avec le oxo

Merci Jean pour ta réponse. J'ai l'erreur 404 pour chaque numero composé, que ce soit un numero interne ou externe, je ne sais pas trop ou chercher dans l'OXE et surtout comment chercher de ce coté la... aurais tu quelques points a me soumettre pour verification?

olppp
29/09/2016, 10h33
bonjour,
Peux-être la nécessité d'un préfixe sur l'oxo ?

bwen
29/09/2016, 12h28
bonjour,
Peux-être la nécessité d'un préfixe sur l'oxo ?

En fait, actuellement l'OXE est fonctionnel avec plusieurs media gateway, j'ai recemment implementé un IPBX pour migrer doucement par la apres les avoir interconnectés mais je bloque a ce niveau pour le moment. La partie "prefixe" est deja pleine d'entrées diverses pour l'utilisation de l'OXE mais faut-il que j'en rajoute specifiquement pour le couplage?

bwen
29/09/2016, 12h30
n'est-il pas possible d'avoir des log detaillés coté OXE comme le sip debug coté asterisk? cela permettrait d'avoir peut etre un peu plus de details sur le 404 not found

olppp
29/09/2016, 14h07
En activant les traps snmp sur l'OXE vers un serveur de supervision. http://wiki.monitoring-fr.org/centreon/superviser-oxe-alcatel

bwen
29/09/2016, 14h14
En activant les traps snmp sur l'OXE vers un serveur de supervision. http://wiki.monitoring-fr.org/centreon/superviser-oxe-alcatel

Merci pour le lien, j'ai un nagios sur lequel je pourrais configurer le snmp mais si je ne m'abuse, cela permet juste de verifier la bonne santé de L'OXE mais pas d'avoir le details des logs pour debugger mon probleme SIP?

bwen
29/09/2016, 15h29
Je progresse doucement, j'ai revu les routage ARS et maintenant le 404 not found s'est transformé en SIP/2.0 480 Temporarily not available.. je ne suis pas bcp plus avancé mais quelqu'un a peut etre une idée?