La trace SIP ne montre rien. Il te manque la trace de CLI pendant l'appel entrant, colle la ici.
J'ai supprimé mon dernier message puisque tu as déjà le renvoie vers numéro correct.
Kalivev1 et 2 est connecté ?
La trace SIP ne montre rien. Il te manque la trace de CLI pendant l'appel entrant, colle la ici.
J'ai supprimé mon dernier message puisque tu as déjà le renvoie vers numéro correct.
Kalivev1 et 2 est connecté ?
Bonjour,
oui, les deux téléphone kalidev 1 et 2 sont connectés. Dans la CLI je met la verbose à 10 mais je n'ai rien qui s'affiche lors d'un appel entrant, comme si l'appel n'atteignait pas mon serveur.
Code:prodserver*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip.ovh.net:5060 N 0033535541xx 105 Registered Tue, 22 Mar 2011 09:01:34 sip.ovh.net:5060 N 0033535541yy 105 Registered Tue, 22 Mar 2011 09:02:37Pour un appel sortant par contre j'ai bien les messages :Code:prodserver*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status forfait-ovh/0033535541xxx 91.121.129.17 N 5060 OK (61 ms) forfait-ovh2/003353554yyy 91.121.129.17 N 5060 OK (60 ms) kalidev1/kalidev1 192.168.1.20 D N 5060 Unmonitored kalidev2/kalidev2 192.168.1.20 D N 5060 Unmonitored 4 sip peers [Monitored: 2 online, 0 offline Unmonitored: 2 online, 0 offline]
Merci de ton aideCode:Verbosity was 5 and is now 10 == Using SIP RTP CoS mark 5 -- Executing [06731705xx@appel-sortant:1] Dial("SIP/kalidev1-0000000a", "SIP/06731705xx@forfait-ovh") in new stack == Using SIP RTP CoS mark 5 -- Called 06731705xx@forfait-ovh -- SIP/forfait-ovh-0000000b is ringing -- SIP/forfait-ovh-0000000b is making progress passing it to SIP/kalidev1-0000000a == Spawn extension (appel-sortant, 06731705xx, 1) exited non-zero on 'SIP/kalidev1-0000000a'
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>