Alors active le sip debug sur le téléphone concerné, et sur tes fournisseurs, appelle et colle tout sur pastebin, ou dans un fichier et l'attache ici.
Alors active le sip debug sur le téléphone concerné, et sur tes fournisseurs, appelle et colle tout sur pastebin, ou dans un fichier et l'attache ici.
Dernière modification par Reaper ; 22/03/2011 à 11h22.
Colle tout ici je te dis.et sur ton fournisseur
voila ce que j'ai avec sip debug sur le fournisseur et le téléphone concerné :
Code:<--- SIP read from UDP:192.168.1.20:5060 ---> REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7661e6afeb5feb59aa739d1eb44f4d7;rport From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133 To: "kalidev1" <sip:kalidev1@192.168.1.10> Call-ID: 2652774740@192_168_1_20 CSeq: 10999 REGISTER Contact: <sip:kalidev1@192.168.1.20:5060> Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.20 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7661e6afeb5feb59aa739d1eb44f4d7;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133 To: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=as79e3afc6 Call-ID: 2652774740@192_168_1_20 CSeq: 10999 REGISTER Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62adcae5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2652774740@192_168_1_20' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.20:5060 ---> REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK6cb7fd43a5a990958942e8acefbc9068;rport From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133 To: "kalidev1" <sip:kalidev1@192.168.1.10> Call-ID: 2652774740@192_168_1_20 CSeq: 11000 REGISTER Contact: <sip:kalidev1@192.168.1.20:5060> Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.10", nonce="62adcae5", response="5454a87e3f25dee5e45d95aaacd0e2cc" Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.1.20 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK6cb7fd43a5a990958942e8acefbc9068;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133 To: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=as79e3afc6 Call-ID: 2652774740@192_168_1_20 CSeq: 11000 REGISTER Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 180 Contact: <sip:kalidev1@192.168.1.20:5060>;expires=180 Date: Tue, 22 Mar 2011 09:44:48 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2652774740@192_168_1_20' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.20:5060 ---> REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3e8ec846d6facef21f0a2dd424d323d9;rport From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850 To: "kalidev 2" <sip:kalidev2@192.168.1.10> Call-ID: 1219423529@192_168_1_20 CSeq: 13700 REGISTER Contact: <sip:kalidev2@192.168.1.20:5060> Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.20 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3e8ec846d6facef21f0a2dd424d323d9;received=192.168.1.20;rport=5060 From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850 To: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=as7b678c2d Call-ID: 1219423529@192_168_1_20 CSeq: 13700 REGISTER Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15f2f0d7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1219423529@192_168_1_20' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.20:5060 ---> REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK74c89957c85cc131509a64262982de0e;rport From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850 To: "kalidev 2" <sip:kalidev2@192.168.1.10> Call-ID: 1219423529@192_168_1_20 CSeq: 13701 REGISTER Contact: <sip:kalidev2@192.168.1.20:5060> Authorization: Digest username="kalidev2", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.10", nonce="15f2f0d7", response="6a14cf5a06890be3d48d57d27ac466d6" Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.1.20 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK74c89957c85cc131509a64262982de0e;received=192.168.1.20;rport=5060 From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850 To: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=as7b678c2d Call-ID: 1219423529@192_168_1_20 CSeq: 13701 REGISTER Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 180 Contact: <sip:kalidev2@192.168.1.20:5060>;expires=180 Date: Tue, 22 Mar 2011 09:44:48 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1219423529@192_168_1_20' in 32000 ms (Method: REGISTER) prodserver*CLI>
voila ce que j'ai si je passe un appel :
Code:<--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f Call-ID: 3641468165@192_168_1_20 CSeq: 3 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:06731705xx@192.168.1.10> Content-Length: 0 <------------> Audio is at 192.168.1.10 port 13190 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f Call-ID: 3641468165@192_168_1_20 CSeq: 3 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:06731705xx@192.168.1.10> Content-Type: application/sdp Content-Length: 260 v=0 o=root 152311073 152311073 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.2.9-2 c=IN IP4 192.168.1.10 t=0 0 m=audio 13190 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.1.20:5060 ---> CANCEL sip:06731705xx@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone Call-ID: 3641468165@192_168_1_20 CSeq: 3 CANCEL Contact: <sip:kalidev1@192.168.1.20:5060> Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="41d4678f083049f066476f80d4b380e8" Max-Forwards: 70 User-Agent: A580 IP021920000000 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.20 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f Call-ID: 3641468165@192_168_1_20 CSeq: 3 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.1.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060 From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f Call-ID: 3641468165@192_168_1_20 CSeq: 3 CANCEL Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:192.168.1.20:5060 ---> ACK sip:06731705xx@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539 To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f Call-ID: 3641468165@192_168_1_20 CSeq: 3 ACK Contact: <sip:kalidev1@192.168.1.20:5060> Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="fdea256982ac7f340489f95dfba9f231" Max-Forwards: 70 User-Agent: A580 IP021920000000 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3641468165@192_168_1_20' Method: ACK [Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:11528 sip_reregister: -- Re-registration for 0033535541287@sip.ovh.net [Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:18270 handle_response_register: Outbound Registration: Expiry for sip.ovh.net is 120 sec (Scheduling reregistration in 105 s) prodserver*CLI>
Bonjour, la trace est incomplète, colle les traces sur pastebin (again)
Et chaque fois il faut spécifier si c'est un appel sortant ou entrant.
En tout cas on vois que c'est le 192.168.1.20 qui termine le communication.
Je vous demande les traces complètes de l'appel qui ne fonctionne pas pour la dernière fois.
désolé je pense que je comprend pas ce dont vous avez besoin, je vous explique ma démarche dite moi ce que je fais de mal (ou ce que je ne fais pas bien sur):
J'ai fait un sip set debug sur mon fournisseur ainsi que mon téléphone
les traces s'affiche sur ma console, je passe un appel (entrant) sur mon téléphone et effectue un copier coller de tous les message à l'écran (#7) (chose qui n'est pas aisé puisque putty à une taille d'écran limité et que les messages continue d'arriver) les messages qui arrivent sont toujours les mêmes et ils me semblent sont des connexions au serveur ovh ... mais je n'ai rien de plus lors d'un appel entrant.
C'est pourquoi ensuite j'ai copier coller la trace d'un appel sortant (#8) pour confirmer que j'étais au bonne endroit et que j'effectuai la bonne manipulation, (c'est bien moi qui est raccrocher avant de décrocher avec l'autre téléphone).
Il y a t il une autre façon de faire que tu copier coller de la CLI j'ai cherché mais je n'ai pas trouvé la façon de créer un fichier avec les messages affichés.
J'essaie vraiment de faire de mon mieux pour vous fournir ce qui pourrait vous servir à m'aider.
Encore merci de me consacrer du temps.