donc, l'appel sort jusqu'"Ã OVH...... un sip set debug on devrait en dire plus
donc, l'appel sort jusqu'"Ã OVH...... un sip set debug on devrait en dire plus
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
J'effectue cette commande sur l'ip de ma machine ? Car quand je tape sip set debug la commande n'est pas reconnue cependant en tapant core show help sip set debug j'ai :
sip set debug {on|off|ip|peer} -- Enable/Desable SIP debugging .
puis une fois les traces obtenues,sip set debug on
sip set debug off
(depuis la cli asterisk)
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
sip set debug on
SIP Debugging enabled
[Dec 11 15:18:06]
[Dec 11 15:18:06] <--- SIP read from UDP:91.121.129.23:5060 --->
[Dec 11 15:18:06] OPTIONS sip:s@192.168.10.65:5060 SIP/2.0
[Dec 11 15:18:06] Call-ID: 06-02130-0a635d93-46c729595@91.121.129.23
[Dec 11 15:18:06] Contact: <sip:91.121.129.23:5060>
[Dec 11 15:18:06] CSeq: 1 OPTIONS
[Dec 11 15:18:06] From: <sip:keepalive@91.121.129.23:5060>;tag=06-02130-0a635d92-011feffb0
[Dec 11 15:18:06] Max-Forwards: 70
[Dec 11 15:18:06] To: <sip:0033...@siptrunk.ovh.net>
[Dec 11 15:18:06] Via: SIP/2.0/UDP 91.121.129.23:5060;rport;branch=z9hG4bK-MTKM-05f4a0f2-0ec87f8e
[Dec 11 15:18:06] Content-Length: 0
[Dec 11 15:18:06]
[Dec 11 15:18:06]
[Dec 11 15:18:06] <------------->
[Dec 11 15:18:06] --- (9 headers 0 lines) ---
[Dec 11 15:18:06] Sending to 91.121.129.23:5060 (no NAT)
[Dec 11 15:18:06] Looking for s in default (domain 192.168.10.65)
[Dec 11 15:18:06]
[Dec 11 15:18:06] <--- Transmitting (no NAT) to 91.121.129.23:5060 --->
[Dec 11 15:18:06] SIP/2.0 404 Not Found
[Dec 11 15:18:06] Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-MTKM-05f4a0f2-0ec87f8e;received=91.121.129.23;rport=5060
[Dec 11 15:18:06] From: <sip:keepalive@91.121.129.23:5060>;tag=06-02130-0a635d92-011feffb0
[Dec 11 15:18:06] To: <sip:0033...@siptrunk.ovh.net>;tag=as07df1f63
[Dec 11 15:18:06] Call-ID: 06-02130-0a635d93-46c729595@91.121.129.23
[Dec 11 15:18:06] CSeq: 1 OPTIONS
[Dec 11 15:18:06] Server: XiVO PBX
[Dec 11 15:18:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 11 15:18:06] Supported: replaces, timer
[Dec 11 15:18:06] Accept: application/sdp
[Dec 11 15:18:06] Content-Length: 0
[Dec 11 15:18:06]
[Dec 11 15:18:06]
[Dec 11 15:18:06] <------------>
[Dec 11 15:18:06] Scheduling destruction of SIP dialog '06-02130-0a635d93-46c729595@91.121.129.23' in 32000 ms (Method: OPTIONS)
[Dec 11 15:18:15]
[Dec 11 15:18:15] <--- SIP read from UDP:192.168.10.62:5178 --->
[Dec 11 15:18:15] SUBSCRIBE sip:23mgt2@192.168.10.65:5060 SIP/2.0
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.62:5178;branch=z9hG4bKfb1ecb08b8b0190d1 10a2c5a5e6d2270;rport
[Dec 11 15:18:15] From: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=494262242
[Dec 11 15:18:15] To: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 484965469 SUBSCRIBE
[Dec 11 15:18:15] Contact: <sip:23mgt2@192.168.10.62:5178>
[Dec 11 15:18:15] Authorization: Digest username="23mgt2", realm="xivo", algorithm=MD5, uri="sip:23mgt2@192.168.10.65:5060", nonce="6c73b7bb", response="3e6d2e0b93df14c7efa684b7c7819652"
[Dec 11 15:18:15] Max-Forwards: 70
[Dec 11 15:18:15] User-Agent: C590 IP/42.075.00.000.000
[Dec 11 15:18:15] Event: message-summary
[Dec 11 15:18:15] Expires: 3600
[Dec 11 15:18:15] Allow: NOTIFY
[Dec 11 15:18:15] Accept: application/simple-message-summary
[Dec 11 15:18:15] Content-Length: 0
[Dec 11 15:18:15]
[Dec 11 15:18:15]
[Dec 11 15:18:15] <------------->
[Dec 11 15:18:15] --- (15 headers 0 lines) ---
[Dec 11 15:18:15] Found peer '23mgt2' for '23mgt2' from 192.168.10.62:5178
[Dec 11 15:18:15] NOTICE[4799]: chan_sip.c:16752 check_auth: Correct auth, but based on stale nonce received from '"23mgt2" <sip:23mgt2@192.168.10.65>;tag=494262242'
[Dec 11 15:18:15]
[Dec 11 15:18:15] <--- Transmitting (no NAT) to 192.168.10.62:5178 --->
[Dec 11 15:18:15] SIP/2.0 401 Unauthorized
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.62:5178;branch=z9hG4bKfb1ecb08b8b0190d1 10a2c5a5e6d2270;received=192.168.10.62;rport=5178
[Dec 11 15:18:15] From: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=494262242
[Dec 11 15:18:15] To: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 484965469 SUBSCRIBE
[Dec 11 15:18:15] Server: XiVO PBX
[Dec 11 15:18:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 11 15:18:15] Supported: replaces, timer
[Dec 11 15:18:15] WWW-Authenticate: Digest algorithm=MD5, realm="xivo", nonce="4ba6c4c8", stale=true
[Dec 11 15:18:15] Content-Length: 0
[Dec 11 15:18:15]
[Dec 11 15:18:15]
[Dec 11 15:18:15] <------------>
[Dec 11 15:18:15] Scheduling destruction of SIP dialog '2516410218@192_168_10_62' in 32000 ms (Method: SUBSCRIBE)
[Dec 11 15:18:15]
[Dec 11 15:18:15] <--- SIP read from UDP:192.168.10.62:5178 --->
[Dec 11 15:18:15] SUBSCRIBE sip:23mgt2@192.168.10.65:5060 SIP/2.0
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.62:5178;branch=z9hG4bK7dea1e4a53dc675fb b5f0ed257b914a0;rport
[Dec 11 15:18:15] From: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=494262242
[Dec 11 15:18:15] To: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 484965470 SUBSCRIBE
[Dec 11 15:18:15] Contact: <sip:23mgt2@192.168.10.62:5178>
[Dec 11 15:18:15] Authorization: Digest username="23mgt2", realm="xivo", algorithm=MD5, uri="sip:23mgt2@192.168.10.65:5060", nonce="4ba6c4c8", response="db27ee34b50177c2000c014633c11023"
[Dec 11 15:18:15] Max-Forwards: 70
[Dec 11 15:18:15] User-Agent: C590 IP/42.075.00.000.000
[Dec 11 15:18:15] Event: message-summary
[Dec 11 15:18:15] Expires: 3600
[Dec 11 15:18:15] Allow: NOTIFY
[Dec 11 15:18:15] Accept: application/simple-message-summary
[Dec 11 15:18:15] Content-Length: 0
[Dec 11 15:18:15]
[Dec 11 15:18:15]
[Dec 11 15:18:15] <------------->
[Dec 11 15:18:15] --- (15 headers 0 lines) ---
[Dec 11 15:18:15] Found peer '23mgt2' for '23mgt2' from 192.168.10.62:5178
[Dec 11 15:18:15] Scheduling destruction of SIP dialog '2516410218@192_168_10_62' in 3610000 ms (Method: SUBSCRIBE)
[Dec 11 15:18:15]
[Dec 11 15:18:15] <--- Transmitting (no NAT) to 192.168.10.62:5178 --->
[Dec 11 15:18:15] SIP/2.0 200 OK
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.62:5178;branch=z9hG4bK7dea1e4a53dc675fb b5f0ed257b914a0;received=192.168.10.62;rport=5178
[Dec 11 15:18:15] From: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=494262242
[Dec 11 15:18:15] To: "23mgt2" <sip:23mgt2@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 484965470 SUBSCRIBE
[Dec 11 15:18:15] Server: XiVO PBX
[Dec 11 15:18:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 11 15:18:15] Supported: replaces, timer
[Dec 11 15:18:15] Expires: 3600
[Dec 11 15:18:15] Contact: <sip:xivo@192.168.10.65:5060>;expires=3600
[Dec 11 15:18:15] Content-Length: 0
[Dec 11 15:18:15]
[Dec 11 15:18:15]
[Dec 11 15:18:15] <------------>
[Dec 11 15:18:15] Reliably Transmitting (no NAT) to 192.168.10.62:5178:
[Dec 11 15:18:15] NOTIFY sip:23mgt2@192.168.10.62:5178 SIP/2.0
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.65:5060;branch=z9hG4bK54a7f562;rport
[Dec 11 15:18:15] Max-Forwards: 70
[Dec 11 15:18:15] Route: <sip:23mgt2@192.168.10.62:5178>
[Dec 11 15:18:15] From: "xivo" <sip:xivo@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] To: <sip:23mgt2@192.168.10.62:5178>;tag=494262242
[Dec 11 15:18:15] Contact: <sip:xivo@192.168.10.65:5060>
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 146 NOTIFY
[Dec 11 15:18:15] User-Agent: XiVO PBX
[Dec 11 15:18:15] Event: message-summary
[Dec 11 15:18:15] Content-Type: application/simple-message-summary
[Dec 11 15:18:15] Subscription-State: active
[Dec 11 15:18:15] Content-Length: 88
[Dec 11 15:18:15]
[Dec 11 15:18:15] Messages-Waiting: no
[Dec 11 15:18:15] Message-Account: sip:*98@192.168.10.65
[Dec 11 15:18:15] Voice-Message: 0/1 (0/0)
[Dec 11 15:18:15]
[Dec 11 15:18:15] ---
[Dec 11 15:18:15]
[Dec 11 15:18:15] <--- SIP read from UDP:192.168.10.62:5178 --->
[Dec 11 15:18:15] SIP/2.0 200 OK
[Dec 11 15:18:15] Via: SIP/2.0/UDP 192.168.10.65:5060;branch=z9hG4bK54a7f562;rport=50 60
[Dec 11 15:18:15] From: "xivo" <sip:xivo@192.168.10.65>;tag=as2a8098af
[Dec 11 15:18:15] To: <sip:23mgt2@192.168.10.62:5178>;tag=494262242
[Dec 11 15:18:15] Call-ID: 2516410218@192_168_10_62
[Dec 11 15:18:15] CSeq: 146 NOTIFY
[Dec 11 15:18:15] User-Agent: C590 IP/42.075.00.000.000
[Dec 11 15:18:15] Content-Length: 0
[Dec 11 15:18:15]
[Dec 11 15:18:15]
[Dec 11 15:18:15] <------------->
[Dec 11 15:18:15] --- (8 headers 0 lines) ---
[Dec 11 15:18:20] == Manager 'xivo_monit_user' logged on from 127.0.0.1
[Dec 11 15:18:25] == Manager 'xivo_monit_user' logged off from 127.0.0.1
[Dec 11 15:18:38] Really destroying SIP dialog '06-02130-0a635d93-46c729595@91.121.129.23' Method: OPTIONS
xivo*CLI> sip set debug off
SIP Debugging Disabled
je ne vois pas d'appel, il faudrait le
core set verbose 3
sip set debug on
lancer l'appel et copier le résultat via pastebin.com !
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
[QUOTE=jean;20034]je ne vois pas d'appel, il faudrait le
core set verbose 3
sip set debug on
lancer l'appel et copier le r
http://pastebin.com/
ca permet de partager des logs, etc... sans polluer le forum - tu copies tes logs sur ce site, et poste le lien ici
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
Salut jean , voila le lien pastebin de mes logs :
http://pastebin.com/9EcxuD0M
J'ai donc fait :
core set verbose 3
sip set debug on
puis j'ai appelé le 06...
Par contre je n'ai pas pu copier plus . Et le lien s'expire dans 1h , je pourrai le remettre .
Merci
Dernière modification par AZ12 ; 14/12/2015 à 12h28.
too late....
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son