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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 0
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voici un extrait de mon fichier de log si ça peut t'aider ?
-------------------------------
---
| Code: |
[Jul 21 18:40:59] VERBOSE[4793] logger.c:
<--- SIP read from 217.64.49.4:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.64.55.158:5060;branch=z9hG4bK76c9fabc;rport;received=62.39.193.70
From: "Unknown" <sip:Unknown@217.64.155.158>;tag=as45a650f4
To: <sip:217.64.49.4>;tag=as78d9b81c
Call-ID: 32ac2b5928520c7f2fede16f734ea1c1@217.64.55.158
CSeq: 102 OPTIONS
User-Agent: PROXY
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:217.64.49.4>
Accept: application/sdp
Content-Length: 0
<------------->
[Jul 21 18:40:59] VERBOSE[4793] logger.c: --- (11 headers 0 lines) ---
[Jul 21 18:40:59] VERBOSE[4793] logger.c: Really destroying SIP dialog '32ac2b5928520c7f2fede16f734ea1c1@217.64.55.158'
Method: OPTIONS
[Jul 21 18:41:26] VERBOSE[4793] logger.c: Reliably Transmitting (NAT) to 217.64.149.4:5060:
OPTIONS sip:217.64.49.4 SIP/2.0
Via: SIP/2.0/UDP 217.64.55.158:5060;branch=z9hG4bK116f7153;rport
From: "Unknown" <sip:Unknown@217.64.55.158>;tag=as70331355
To: <sip:217.64.49.4>
Contact: <sip:Unknown@217.64.55.158>
Call-ID: 09904c36511ff78e1b1714813632c508@217.64.55.158
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Jul 2009 16:41:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Jul 21 18:41:26] VERBOSE[4793] logger.c:
<--- SIP read from 217.64.49.4:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.64.155.158:5060;branch=z9hG4bK116f7153;rport;received=62.39.193.70
From: "Unknown" <sip:Unknown@217.64.155.158>;tag=as70331355
To: <sip:217.64.149.4>;tag=as3bb2c612
Call-ID: 09904c36511ff78e1b1714813632c508@217.64.155.158
CSeq: 102 OPTIONS
User-Agent: PROXY
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:217.64.49.4>
Accept: application/sdp
Content-Length: 0
<------------->
[Jul 21 18:41:26] VERBOSE[4793] logger.c: --- (11 headers 0 lines) ---
[Jul 21 18:41:26] VERBOSE[4793] logger.c: Really destroying SIP dialog '09904c36511ff78e1b1714813632c508@217.64.155.158'
Method: OPTIONS
[Jul 21 18:41:28] VERBOSE[5122] logger.c: Scheduling destruction of SIP dialog '18ea58bc30f4828c1384bc553b526113@217.64
.155.158' in 6400 ms (Method: INVITE)
[Jul 21 18:41:28] DEBUG[5122] chan_sip.c: Strict routing enforced for session 18ea58bc30f4828c1384bc553b526113@217.64.5
5.158
[Jul 21 18:41:28] VERBOSE[5122] logger.c: set_destination: Parsing <sip:0811851851@217.64.49.4> for address/port to sen
d to
[Jul 21 18:41:28] VERBOSE[5122] logger.c: set_destination: set destination to 217.64.149.4, port 5060
[Jul 21 18:41:28] VERBOSE[5122] logger.c: Reliably Transmitting (NAT) to 217.64.149.4:5060:
BYE sip:0811851851@217.64.149.4 SIP/2.0
Via: SIP/2.0/UDP 217.64.155.158:5060;branch=z9hG4bK70a9d78b;rport
From: "0489614120" <sip:0489614120@217.64.55.158>;tag=as00cd085d
To: <sip:0811851851@217.64.49.4>;tag=as0c23737e
Call-ID: 18ea58bc30f4828c1384bc553b526113@217.64.55.158
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[Jul 21 18:41:28] VERBOSE[5122] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/101
-082907e8' in macro 'dialout-trunk'
[Jul 21 18:41:28] VERBOSE[5122] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/101
-082907e8'
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/101-082907e8", "hangupc
all|") in new stack
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/101-082907e8", "w") in
new stack
[Jul 21 18:41:28] DEBUG[5122] app_macro.c: Executed application: ResetCDR
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/101-082907e8", "") in new
stack
[Jul 21 18:41:28] DEBUG[5122] app_macro.c: Executed application: NoCDR
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/101-082907e8", "1?skiprg"
) in new stack
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Goto (macro-hangupcall,s,6)
[Jul 21 18:41:28] DEBUG[5122] app_macro.c: Executed application: GotoIf
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/101-082907e8", "1?skipblk
vm") in new stack
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Goto (macro-hangupcall,s,9)
[Jul 21 18:41:28] DEBUG[5122] app_macro.c: Executed application: GotoIf
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/101-082907e8", "1?theend"
) in new stack
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Goto (macro-hangupcall,s,11)
[Jul 21 18:41:28] DEBUG[5122] app_macro.c: Executed application: GotoIf
[Jul 21 18:41:28] VERBOSE[5122] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/101-082907e8", "") in ne
w stack
[Jul 21 18:41:28] VERBOSE[5122] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/101-08
2907e8' in macro 'hangupcall'
[Jul 21 18:41:28] VERBOSE[5122] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/101-08
2907e8'
[Jul 21 18:41:28] DEBUG[5122] chan_sip.c: Call from peer '101' removed from call limit 50
[Jul 21 18:41:28] VERBOSE[4793] logger.c:
<--- SIP read from 217.64.149.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.155.158:5060;branch=z9hG4bK70a9d78b;received=62.39.193.70;rport=1024
From: "0489614120" <sip:0489614120@217.64.155.158>;tag=as00cd085d
To: <sip:0811851851@217.64.49.4>;tag=as0c23737e
Call-ID: 18ea58bc30f4828c1384bc553b526113@217.64.155.158
CSeq: 103 BYE
User-Agent: PROXY
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0811851851@217.64.149.4>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
|
------------------------------------
Après vérification, notre opérateur envoie bien notre n° sur 11 chiffres :334XXXXXXXX
------------------------------------ |
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Last Edit: 2009/07/21 12:15 By danardf.
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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 162
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Regardes si dans les paramètres généraux Allow Anonymous Inbound SIP Calls est à yes.
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Links hidden for unregistered users. Login or register Here - Links hidden for unregistered users. Login or register Here - Franck Danard - franck.danard@roomx.fr |
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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 0
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YES!!!! ça marche, en effet anonymous SIP calls était sur NO...
Merci INFINIMENT on va pouvoir continuer à se faire harceler par les clients maintenant  |
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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 162
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Harceler les clients à cette heure?
Vite que je te black list 
Quand tu regardes la trace , tu as un :
[Jul 21 18:41:26] VERBOSE[4793] logger.c:
<--- SIP read from 217.64.49.4:5060 --->
SIP/2.0 404 Not Found
Qu'elle est le malotru qui a toucher à çà?  |
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Last Edit: 2009/07/21 12:35 By danardf.
Links hidden for unregistered users. Login or register Here - Links hidden for unregistered users. Login or register Here - Franck Danard - franck.danard@roomx.fr |
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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 0
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SE faire harceler, car les appels sortants passaient bien, c'étaient les entrants qui n'entraient plus
Merci encore pour ta patience  (à charge de revanche dans un autre domaine (l'alcool ? ) |
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Re:plus d'appels entrants... 3 Years, 11 Months ago
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Karma: 162
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L'alcool.
Ben si tu peux avoir du Rhum de Guadeloupe en BIB à 59°, pourquoi pas   |
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Links hidden for unregistered users. Login or register Here - Links hidden for unregistered users. Login or register Here - Franck Danard - franck.danard@roomx.fr |
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