XIVO avec SPA3102 + FREEBOX Pb en sortie

Version imprimable

13/07/2015, 20h28
jany1975

XIVO avec SPA3102 + FREEBOX Pb en sortie

Bonjour Messieurs Dames,

J'ai installé un Routeur VOIP SPA3102 au niveau de ma Freebox pour m'amuser un peu.
J'ai fait le tour du tour sur internet et impossible de trouver une solution...

Je reçois les appels sans problème mais impossible d'en passer la ligne apparait comme occupée.

voici les infos d'asterisk.

************************************************** *****************************
[Jul 13 20:11:48] -- Executing [0683158116@default:1] Set("SCCP/105-00000062", "XIVO_BASE_CONTEXT=default") in new stack
[Jul 13 20:11:48] -- Executing [0683158116@default:2] Set("SCCP/105-00000062", "XIVO_BASE_EXTEN=0123456789") in new stack
[Jul 13 20:11:48] -- Executing [0683158116@default:3] Gosub("SCCP/105-00000062", "outcall,s,1(5,)") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:1] Set("SCCP/105-00000062", "XIVO_DSTID=5") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:2] Set("SCCP/105-00000062", "XIVO_PRESUBR_GLOBAL_NAME=OUTCALL") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:3] Set("SCCP/105-00000062", "XIVO_SRCNUM=105") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:4] Set("SCCP/105-00000062", "XIVO_DSTNUM=123456789") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:5] Set("SCCP/105-00000062", "XIVO_CONTEXT=default") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:6] Gosub("SCCP/105-00000062", "originate-caller-id,s,1") in new stack
[Jul 13 20:11:48] -- Executing [s@originate-caller-id:1] GotoIf("SCCP/105-00000062", "0?:name") in new stack
[Jul 13 20:11:48] -- Goto (originate-caller-id,s,3)
[Jul 13 20:11:48] -- Executing [s@originate-caller-id:3] GotoIf("SCCP/105-00000062", "0?:end") in new stack
[Jul 13 20:11:48] -- Goto (originate-caller-id,s,5)
[Jul 13 20:11:48] -- Executing [s@originate-caller-id:5] Return("SCCP/105-00000062", "") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:7] AGI("SCCP/105-00000062", "agi://127.0.0.1/outgoing_user_set_features") in new stack
[Jul 13 20:11:48] agi://127.0.0.1/outgoing_user_set_features: AGI handler 'outgoing_user_set_features' successfully executed
[Jul 13 20:11:48] -- <SCCP/105-00000062>AGI Script agi://127.0.0.1/outgoing_user_set_features completed, returning 0
[Jul 13 20:11:48] -- Executing [s@outcall:8] Gosub("SCCP/105-00000062", "xivo-subroutine,s,1()") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-subroutine:1] GotoIf("SCCP/105-00000062", "?:nosubroutine") in new stack
[Jul 13 20:11:48] -- Goto (xivo-subroutine,s,4)
[Jul 13 20:11:48] -- Executing [s@xivo-subroutine:4] Return("SCCP/105-00000062", "") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:9] Gosub("SCCP/105-00000062", "xivo-user_rights_check,s,1") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-user_rights_check:1] AGI("SCCP/105-00000062", "agi://127.0.0.1/user_set_call_rights") in new stack
[Jul 13 20:11:48] agi://127.0.0.1/user_set_call_rights: AGI handler 'user_set_call_rights' successfully executed
[Jul 13 20:11:48] -- <SCCP/105-00000062>AGI Script agi://127.0.0.1/user_set_call_rights completed, returning 0
[Jul 13 20:11:48] -- Executing [s@xivo-user_rights_check:2] GotoIf("SCCP/105-00000062", "ALLOW?:error,1") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-user_rights_check:3] GotoIf("SCCP/105-00000062", "1?allow,1") in new stack
[Jul 13 20:11:48] -- Goto (xivo-user_rights_check,allow,1)
[Jul 13 20:11:48] -- Executing [allow@xivo-user_rights_check:1] NoOp("SCCP/105-00000062", "User allowed to make call") in new stack
[Jul 13 20:11:48] -- Executing [allow@xivo-user_rights_check:2] Return("SCCP/105-00000062", "") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:10] AGI("SCCP/105-00000062", "agi://127.0.0.1/check_schedule") in new stack
[Jul 13 20:11:48] agi://127.0.0.1/check_schedule: AGI handler 'check_schedule' successfully executed
[Jul 13 20:11:48] -- <SCCP/105-00000062>AGI Script agi://127.0.0.1/check_schedule completed, returning 0
[Jul 13 20:11:48] -- Executing [s@outcall:11] GotoIf("SCCP/105-00000062", "0?CLOSED,1") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:12] GotoIf("SCCP/105-00000062", "?:14") in new stack
[Jul 13 20:11:48] -- Goto (outcall,s,14)
[Jul 13 20:11:48] -- Executing [s@outcall:14] GotoIf("SCCP/105-00000062", "SIP/rtc?:error,1") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:15] Set("SCCP/105-00000062", "TRUNKINDEX=0") in new stack
[Jul 13 20:11:48] -- Executing [s@outcall:16] Goto("SCCP/105-00000062", "dial,1") in new stack
[Jul 13 20:11:48] -- Goto (outcall,dial,1)
[Jul 13 20:11:48] -- Executing [dial@outcall:1] Set("SCCP/105-00000062", "INTERFACE=SIP/rtc") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:2] Set("SCCP/105-00000062", "TRUNKEXTEN=0123456789") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:3] Set("SCCP/105-00000062", "TRUNKSUFFIX=") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:4] Gosub("SCCP/105-00000062", "xivo-global-subroutine,s,1") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-global-subroutine:1] GotoIf("SCCP/105-00000062", "1?:return") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-global-subroutine:2] GotoIf("SCCP/105-00000062", "OUTCALL?:return") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-global-subroutine:3] GotoIf("SCCP/105-00000062", "xivo-subrgbl-outcall?:return") in new stack
[Jul 13 20:11:48] -- Executing [s@xivo-global-subroutine:4] GotoIf("SCCP/105-00000062", "0?:return") in new stack
[Jul 13 20:11:48] -- Goto (xivo-global-subroutine,s,6)
[Jul 13 20:11:48] -- Executing [s@xivo-global-subroutine:6] Return("SCCP/105-00000062", "") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:5] CELGenUserEvent("SCCP/105-00000062", "XIVO_OUTCALL") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:6] Set("SCCP/105-00000062", "CONNECTEDLINE(num,i)=0123456789") in new stack
[Jul 13 20:11:48] -- Executing [dial@outcall:7] Dial("SCCP/105-00000062", "SIP/rtc/0683158116,,o(0123456789)T") in new stack
[Jul 13 20:11:48] == Using SIP RTP CoS mark 5
[Jul 13 20:11:48] -- Called SIP/rtc/0123456789
[Jul 13 20:11:48] == Everyone is busy/congested at this time (1:0/0/1)
[Jul 13 20:11:48] -- Executing [dial@outcall:8] Goto("SCCP/105-00000062", "CHANUNAVAIL,1") in new stack
[Jul 13 20:11:48] -- Goto (outcall,CHANUNAVAIL,1)
[Jul 13 20:11:48] -- Executing [CHANUNAVAIL@outcall:1] Goto("SCCP/105-00000062", "redial,1") in new stack
[Jul 13 20:11:48] -- Goto (outcall,redial,1)
[Jul 13 20:11:48] -- Executing [redial@outcall:1] Set("SCCP/105-00000062", "TRUNKINDEX=1") in new stack
[Jul 13 20:11:48] -- Executing [redial@outcall:2] GotoIf("SCCP/105-00000062", "?dial,1") in new stack
[Jul 13 20:11:48] -- Executing [redial@outcall:3] Playback("SCCP/105-00000062", "congestion-call") in new stack
[Jul 13 20:11:48] == Using sccp rtp TOS bits 184
[Jul 13 20:11:48] > 0x98fefb0 -- Probation passed - setting RTP source address to 192.168.5.100:31888
[Jul 13 20:11:48] -- <SCCP/105-00000062> Playing 'congestion-call.slin' (language 'fr_FR')
******************************

je trouve bien ce problème Everyone is busy/congested at this time (1:0/0/1) mais je ne comprends pas d'où il vient...

J'ai fait le tour des infos regional, pstn line et malgré toutes les configurations essayées, il m'est toujours impossible de faire sortir mes appels.

Est ce qu'il faut changer quelque chose pour la tonalité rtc avec le freebox ???

config cote routeur:

*****************************************
Call Progress Tones
Dial Tone: 440@-19;10(*/0/1)
Second Dial Tone: 420@-19,520@-19;10(*/0/1+2)
Outside Dial Tone: 420@-16;10(*/0/1)
Prompt Tone: 520@-19,620@-19;10(*/0/1+2)
Busy Tone: 440@-19,440@-19;10(.5/.5/1)
Reorder Tone: 480@-19,620@-19;10(.25/.25/1+2)
Off Hook Warning Tone: 480@-10,620@0;10(.125/.125/1+2)
Ring Back Tone: 440@-19;*(1.5/3.5/1)
Ring Back 2 Tone: 440@-19,480@-19;*(1/1/1+2)
Confirm Tone: 600@-16;1(.25/.25/1)

le reste est standard.

sinon CWT Frequency: 440@-10

************************************************** *

Merci à vous.

Jany.
13/07/2015, 20h42
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.
15/07/2015, 22h31
jany1975

Citation:

Envoyé par 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.

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
15/07/2015, 23h19
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)
19/07/2015, 17h19
jany1975

Citation:

Envoyé par 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)

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.
19/07/2015, 18h32
jany1975

qualify = yes

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.
19/07/2015, 18h59
jany1975

résultat

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
19/07/2015, 19h00
jany1975

suite

[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'
20/07/2015, 14h44
jean

<--- 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>

le boitier répind cela... pourquoi, je ne saurai dire....
23/07/2015, 22h16
jany1975

ça fonctionne

Bonjour Jean, et merci pour ton aide, j'ai fini par refaire tous les paramètres PSTN et sip... là, je pouvais appeler mais impossible de recevoir....
j'ai mis 6 Heures avant de voir que je n'utilisais tout simplement pas le bon dialplan au niveau du spa pour la réception.... :whistle:

Je suis surpris du résultat au niveau du fonctionnement, la qualité semble plutôt sympa et le spa décroche et raccroche vraiment bien sur la Freebox... on verra dans quelques jours mais je suis vraiment agréablement surpris par ce petit boitier.

Dans tous les cas, un grand merci à toi pour tes réponses et ton aide.

Au plaisir. Jany.