XIVO avec SPA3102 + FREEBOX Pb en sortie

[jean]

probablement la box n'est pas enregistrée... sip show peers

le peer 'rtc' ne doit pas être disponible - il faut vérifier que le spa s'enregistre correctement

j.

[jany1975]

probablement la box n'est pas enregistrée... sip show peers

le peer 'rtc' ne doit pas être disponible - il faut vérifier que le spa s'enregistre correctement

j.

Bonjour Jean et merci pour ton aide,

Le spa avec l'utilisateur rtc apparait pourtant bien dans sip show peers ...


Voici mon sip show peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
pltb82/pltb82 (Unspecified) D No No 0 Unmonitored
rtc/rtc 192.168.1.251 No No 5060 Unmonitored
ippi/ippi 213.215.45.230 No No 5060 Unmonitored

Merci

[jean]

rtc n'est pas un hote dynamique, donc le sip show peers est peu concluant
rajoute un qualify=yes pour rtc, et ensuite regarde si il apparait bien OK avec un temps
ensuite, fais un sip set debug ip 192.168.1.251 sur la console, et poste le résultat (éventuellement via pastebin) - mais si il y a du traffic , tu devrais voir qui rejette l'appel (et pourquoi avec un peu de chance)

[jany1975]

rtc n'est pas un hote dynamique, donc le sip show peers est peu concluant
rajoute un qualify=yes pour rtc, et ensuite regarde si il apparait bien OK avec un temps
ensuite, fais un sip set debug ip 192.168.1.251 sur la console, et poste le résultat (éventuellement via pastebin) - mais si il y a du traffic , tu devrais voir qui rejette l'appel (et pourquoi avec un peu de chance)

Bonjour Jean,

J'ai essayé de rajouter le "qualify=yes" dans sip.conf mais xivo efface la ligne au rechargement de la config, est ce qu'il y a un autre moyen d'activer cette option ???

Merci.

Jany.

[jany1975]

J'ai trouvé pour activer le qualify = yes en fait la paramétrage se trouve dans le protocol sip et non la connexion...
Voici la sip show peers qui parait tout bon:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
7kqldv/7kqldv 192.168.1.39 D No No 61393 UNREACHABLE
rtc/rtc 192.168.1.251 No No 5060 OK (6 ms)
ippi/ippi 213.215.45.230 No No 5060 OK (1282 ms)

merci pour votre aide.

Jany.

[jany1975]

Voici le résultat du debug:

*************************************************
[Jul 19 18:50:54] == Using SIP RTP CoS mark 5
[Jul 19 18:50:54] -- Executing [0325710285@default:1] Set("SIP/7kqldv-00000015", "XIVO_BASE_CONTEXT=default") in new stack
[Jul 19 18:50:54] -- Executing [0325710285@default:2] Set("SIP/7kqldv-00000015", "XIVO_BASE_EXTEN=0325710285") in new stack
[Jul 19 18:50:54] -- Executing [0325710285@default:3] Gosub("SIP/7kqldv-00000015", "outcall,s,1(5,)") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:1] Set("SIP/7kqldv-00000015", "XIVO_DSTID=5") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:2] Set("SIP/7kqldv-00000015", "XIVO_PRESUBR_GLOBAL_NAME=OUTCALL") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:3] Set("SIP/7kqldv-00000015", "XIVO_SRCNUM=106") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:4] Set("SIP/7kqldv-00000015", "XIVO_DSTNUM=0325710285") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:5] Set("SIP/7kqldv-00000015", "XIVO_CONTEXT=default") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:6] Gosub("SIP/7kqldv-00000015", "originate-caller-id,s,1") in new stack
[Jul 19 18:50:54] -- Executing [s@originate-caller-id:1] GotoIf("SIP/7kqldv-00000015", "0?:name") in new stack
[Jul 19 18:50:54] -- Goto (originate-caller-id,s,3)
[Jul 19 18:50:54] -- Executing [s@originate-caller-id:3] GotoIf("SIP/7kqldv-00000015", "0?:end") in new stack
[Jul 19 18:50:54] -- Goto (originate-caller-id,s,5)
[Jul 19 18:50:54] -- Executing [s@originate-caller-id:5] Return("SIP/7kqldv-00000015", "") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:7] AGI("SIP/7kqldv-00000015", "agi://127.0.0.1/outgoing_user_set_features") in new stack
[Jul 19 18:50:54] agi://127.0.0.1/outgoing_user_set_features: AGI handler 'outgoing_user_set_features' successfully executed
[Jul 19 18:50:54] -- <SIP/7kqldv-00000015>AGI Script agi://127.0.0.1/outgoing_user_set_features completed, returning 0
[Jul 19 18:50:54] -- Executing [s@outcall:8] Gosub("SIP/7kqldv-00000015", "xivo-subroutine,s,1()") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-subroutine:1] GotoIf("SIP/7kqldv-00000015", "?:nosubroutine") in new stack
[Jul 19 18:50:54] -- Goto (xivo-subroutine,s,4)
[Jul 19 18:50:54] -- Executing [s@xivo-subroutine:4] Return("SIP/7kqldv-00000015", "") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:9] Gosub("SIP/7kqldv-00000015", "xivo-user_rights_check,s,1") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-user_rights_check:1] AGI("SIP/7kqldv-00000015", "agi://127.0.0.1/user_set_call_rights") in new stack
[Jul 19 18:50:54] agi://127.0.0.1/user_set_call_rights: AGI handler 'user_set_call_rights' successfully executed
[Jul 19 18:50:54] -- <SIP/7kqldv-00000015>AGI Script agi://127.0.0.1/user_set_call_rights completed, returning 0
[Jul 19 18:50:54] -- Executing [s@xivo-user_rights_check:2] GotoIf("SIP/7kqldv-00000015", "ALLOW?:error,1") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-user_rights_check:3] GotoIf("SIP/7kqldv-00000015", "1?allow,1") in new stack
[Jul 19 18:50:54] -- Goto (xivo-user_rights_check,allow,1)
[Jul 19 18:50:54] -- Executing [allow@xivo-user_rights_check:1] NoOp("SIP/7kqldv-00000015", "User allowed to make call") in new stack
[Jul 19 18:50:54] -- Executing [allow@xivo-user_rights_check:2] Return("SIP/7kqldv-00000015", "") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:10] AGI("SIP/7kqldv-00000015", "agi://127.0.0.1/check_schedule") in new stack
[Jul 19 18:50:54] agi://127.0.0.1/check_schedule: AGI handler 'check_schedule' successfully executed
[Jul 19 18:50:54] -- <SIP/7kqldv-00000015>AGI Script agi://127.0.0.1/check_schedule completed, returning 0
[Jul 19 18:50:54] -- Executing [s@outcall:11] GotoIf("SIP/7kqldv-00000015", "0?CLOSED,1") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:12] GotoIf("SIP/7kqldv-00000015", "?:14") in new stack
[Jul 19 18:50:54] -- Goto (outcall,s,14)
[Jul 19 18:50:54] -- Executing [s@outcall:14] GotoIf("SIP/7kqldv-00000015", "SIP/rtc?:error,1") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:15] Set("SIP/7kqldv-00000015", "TRUNKINDEX=0") in new stack
[Jul 19 18:50:54] -- Executing [s@outcall:16] Goto("SIP/7kqldv-00000015", "dial,1") in new stack
[Jul 19 18:50:54] -- Goto (outcall,dial,1)
[Jul 19 18:50:54] -- Executing [dial@outcall:1] Set("SIP/7kqldv-00000015", "INTERFACE=SIP/rtc") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:2] Set("SIP/7kqldv-00000015", "TRUNKEXTEN=0325710285") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:3] Set("SIP/7kqldv-00000015", "TRUNKSUFFIX=") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:4] Gosub("SIP/7kqldv-00000015", "xivo-global-subroutine,s,1") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-global-subroutine:1] GotoIf("SIP/7kqldv-00000015", "1?:return") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-global-subroutine:2] GotoIf("SIP/7kqldv-00000015", "OUTCALL?:return") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-global-subroutine:3] GotoIf("SIP/7kqldv-00000015", "xivo-subrgbl-outcall?:return") in new stack
[Jul 19 18:50:54] -- Executing [s@xivo-global-subroutine:4] GotoIf("SIP/7kqldv-00000015", "0?:return") in new stack
[Jul 19 18:50:54] -- Goto (xivo-global-subroutine,s,6)
[Jul 19 18:50:54] -- Executing [s@xivo-global-subroutine:6] Return("SIP/7kqldv-00000015", "") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:5] CELGenUserEvent("SIP/7kqldv-00000015", "XIVO_OUTCALL") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:6] Set("SIP/7kqldv-00000015", "CONNECTEDLINE(num,i)=0325710285") in new stack
[Jul 19 18:50:54] -- Executing [dial@outcall:7] Dial("SIP/7kqldv-00000015", "SIP/rtc/0325710285,,o(0325710285)T") in new stack
[Jul 19 18:50:54] == Using SIP RTP CoS mark 5
[Jul 19 18:50:54] Audio is at 16272
[Jul 19 18:50:54] Adding codec 100004 (alaw) to SDP
[Jul 19 18:50:54] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 19 18:50:54] Reliably Transmitting (no NAT) to 192.168.1.251:5060:
[Jul 19 18:50:54] INVITE sip:0325710285@192.168.1.251:5060 SIP/2.0
[Jul 19 18:50:54] Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK4b1230d6
[Jul 19 18:50:54] Max-Forwards: 70
[Jul 19 18:50:54] From: "JANY GSM" <sip:106@192.168.1.251>;tag=as67a4308e
[Jul 19 18:50:54] To: <sip:0325710285@192.168.1.251:5060>
[Jul 19 18:50:54] Contact: <sip:106@192.168.1.200:5060>
[Jul 19 18:50:54] Call-ID: 39d4a23e6f8b96be144a457439b45f71@192.168.1.251
[Jul 19 18:50:54] CSeq: 102 INVITE
[Jul 19 18:50:54] User-Agent: XiVO PBX
[Jul 19 18:50:54] Date: Sun, 19 Jul 2015 16:50:54 GMT
[Jul 19 18:50:54] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 19 18:50:54] Supported: replaces, timer
[Jul 19 18:50:54] Content-Type: application/sdp
[Jul 19 18:50:54] Content-Length: 281
[Jul 19 18:50:54]
[Jul 19 18:50:54] v=0
[Jul 19 18:50:54] o=root 1926845718 1926845718 IN IP4 192.168.1.200
[Jul 19 18:50:54] s=Asterisk PBX 11.17.1+xivo.15.11~20150608.145923.95f0bc00-wheezy
[Jul 19 18:50:54] c=IN IP4 192.168.1.200
[Jul 19 18:50:54] t=0 0
[Jul 19 18:50:54] m=audio 16272 RTP/AVP 8 101
[Jul 19 18:50:54] a=rtpmap:8 PCMA/8000
[Jul 19 18:50:54] a=rtpmap:101 telephone-event/8000
[Jul 19 18:50:54] a=fmtp:101 0-16
[Jul 19 18:50:54] a=ptime:20
[Jul 19 18:50:54] a=sendrecv
[Jul 19 18:50:54]


***********************************
a suivre

[jany1975]

[Jul 19 18:50:54] ---
[Jul 19 18:50:54] -- Called SIP/rtc/0325710285
[Jul 19 18:50:54]
[Jul 19 18:50:54] <--- SIP read from UDP:192.168.1.251:5060 --->
[Jul 19 18:50:54] SIP/2.0 404 Not Found
[Jul 19 18:50:54] To: <sip:0325710285@192.168.1.251:5060>;tag=1660420a52 85c99ei0
[Jul 19 18:50:54] From: "JANY GSM" <sip:106@192.168.1.251>;tag=as67a4308e
[Jul 19 18:50:54] Call-ID: 39d4a23e6f8b96be144a457439b45f71@192.168.1.251
[Jul 19 18:50:54] CSeq: 102 INVITE
[Jul 19 18:50:54] Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK4b1230d6
[Jul 19 18:50:54] Server: Linksys/SPA3102-5.2.13(GW002)
[Jul 19 18:50:54] Content-Length: 0
[Jul 19 18:50:54]
[Jul 19 18:50:54]
[Jul 19 18:50:54] <------------->
[Jul 19 18:50:54] --- (8 headers 0 lines) ---
[Jul 19 18:50:54] Transmitting (no NAT) to 192.168.1.251:5060:
[Jul 19 18:50:54] ACK sip:0325710285@192.168.1.251:5060 SIP/2.0
[Jul 19 18:50:54] Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK4b1230d6
[Jul 19 18:50:54] Max-Forwards: 70
[Jul 19 18:50:54] From: "JANY GSM" <sip:106@192.168.1.251>;tag=as67a4308e
[Jul 19 18:50:54] To: <sip:0325710285@192.168.1.251:5060>;tag=1660420a52 85c99ei0
[Jul 19 18:50:54] Contact: <sip:106@192.168.1.200:5060>
[Jul 19 18:50:54] Call-ID: 39d4a23e6f8b96be144a457439b45f71@192.168.1.251
[Jul 19 18:50:54] CSeq: 102 ACK
[Jul 19 18:50:54] User-Agent: XiVO PBX
[Jul 19 18:50:54] Content-Length: 0
[Jul 19 18:50:54]
[Jul 19 18:50:54]
[Jul 19 18:50:54] ---
[Jul 19 18:50:54] Scheduling destruction of SIP dialog '39d4a23e6f8b96be144a457439b45f71@192.168.1.251' in 6400 ms (Method: INVITE)
[Jul 19 18:50:54] == Everyone is busy/congested at this time (1:0/0/1)
[Jul 19 18:50:54] -- Executing [dial@outcall:8] Goto("SIP/7kqldv-00000015", "CHANUNAVAIL,1") in new stack
[Jul 19 18:50:54] -- Goto (outcall,CHANUNAVAIL,1)
[Jul 19 18:50:54] -- Executing [CHANUNAVAIL@outcall:1] Goto("SIP/7kqldv-00000015", "redial,1") in new stack
[Jul 19 18:50:54] -- Goto (outcall,redial,1)
[Jul 19 18:50:54] -- Executing [redial@outcall:1] Set("SIP/7kqldv-00000015", "TRUNKINDEX=1") in new stack
[Jul 19 18:50:54] -- Executing [redial@outcall:2] GotoIf("SIP/7kqldv-00000015", "?dial,1") in new stack
[Jul 19 18:50:54] -- Executing [redial@outcall:3] Playback("SIP/7kqldv-00000015", "congestion-call") in new stack
[Jul 19 18:50:55] -- <SIP/7kqldv-00000015> Playing 'congestion-call.slin' (language 'fr_FR')
[Jul 19 18:50:57] > 0x9266398 -- Probation passed - setting RTP source address to 192.168.1.39:51592
[Jul 19 18:50:57]
[Jul 19 18:50:57] <--- SIP read from UDP:192.168.1.251:5060 --->
[Jul 19 18:50:57] NOTIFY sip:192.168.1.200 SIP/2.0
[Jul 19 18:50:57] Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK-16235e3c
[Jul 19 18:50:57] From: rtc <sip:rtc@192.168.1.200>;tag=26b3ecba3f2e9eaeo1
[Jul 19 18:50:57] To: <sip:192.168.1.200>
[Jul 19 18:50:57] Call-ID: 9498754a-43d92f5e@192.168.1.251
[Jul 19 18:50:57] CSeq: 34288 NOTIFY
[Jul 19 18:50:57] Max-Forwards: 70
[Jul 19 18:50:57] Contact: rtc <sip:rtc@192.168.1.251:5060>
[Jul 19 18:50:57] Event: keep-alive
[Jul 19 18:50:57] User-Agent: Linksys/SPA3102-5.2.13(GW002)
[Jul 19 18:50:57] Content-Length: 0
[Jul 19 18:50:57]
[Jul 19 18:50:57]
[Jul 19 18:50:57] <------------->
[Jul 19 18:50:57] --- (11 headers 0 lines) ---
[Jul 19 18:50:57]
[Jul 19 18:50:57] <--- Transmitting (no NAT) to 192.168.1.251:5060 --->
[Jul 19 18:50:57] SIP/2.0 200 OK
[Jul 19 18:50:57] Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK-16235e3c;received=192.168.1.251
[Jul 19 18:50:57] From: rtc <sip:rtc@192.168.1.200>;tag=26b3ecba3f2e9eaeo1
[Jul 19 18:50:57] To: <sip:192.168.1.200>;tag=as513e6eb6
[ Call-ID: 9498754a-43d92f5e@192.168.1.251
CSeq: 34288 NOTIFY
Server: XiVO PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '9498754a-43d92f5e@192.168.1.251' in 32000 ms (Method: NOTIFY)
Really destroying SIP dialog '39d4a23e6f8b96be144a457439b45f71@192.168.1.251' Method: INVITE
-- Executing [redial@outcall:4] Hangup("SIP/7kqldv-00000015", "") in new stack
== Spawn extension (outcall, redial, 4) exited non-zero on 'SIP/7kqldv-00000015'