Bonjour,
Cela fait plusieurs jours que je suis bloqué dans la configuration d'asterisk.
Les appels sortant fonctionne très bien avec mes 2 lignes ovh mais au cours d'un appel entrant sur l'une ou l'autre de mes lignes je n'ai pour seul réponse: "la personne au poste xx n'est pas disponible".
Mon serveur asterisk (1.6.2.9) est derrière une bbox (port 5060 et 10000 à 15000 rediriger).
Voila ma configuration :
sip.conf
extention.confCode:[general] context=forfait-ovh bindport=5060 bindaddr=0.0.0.0 srvlookup=yes register => 003353554xxxx:MDP@sip.ovh.net:5060/003353554xxxx register => 003353554yyyy:MDP@sip.ovh.net:5060/003353554yyyy [kalidev1] type=friend auth=md5 username=kalidev1 md5secret=034093055c683d1f9aee6e49682b04fe callerid="Kalidev 1" <003353554xxxx> host=dynamic context=appel-sortant language=fr insecure=port nat=no canreinvite=no dtmfmode=auto video=no restrictcid=no amaflags=default [kalidev2] type=friend auth=md5 username=kalidev2 md5secret=da61f0014fa319d2ffaebc8472e008a1 callerid="Kalidev 2" <003353554yyyy> host=dynamic context=appel-sortant2 language=fr insecure=port nat=no canreinvite=no dtmfmode=auto video=no restrictcid=no amaflags=default [forfait-ovh] type=peer host=sip.ovh.net fromuser=003353554xxxx fromdomain=sip.ovh.net context=ovh-sip language=fr insecure=port,invite username=003353554xxxx secret=MDP nat=yes canreinvite=no dtmfmode=auto video=no restrictcid=no amaflags=default qualify=yes [forfait-ovh2] type=peer host=sip.ovh.net fromuser=003353554yyyy fromdomain=sip.ovh.net context=ovh-sip language=fr insecure=port,invite username=003353554yyyy secret=MDP nat=yes canreinvite=no dtmfmode=auto video=no restrictcid=no amaflags=default qualify=yes
le debug CLI :Code:[general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no ;[globals] ;CONSOLE=Console/dsp ;IAXINFO=guest ;TRUNK=Zap/g2 ;TRUNKMSD=1 [ovh-sip] ; nom du plan exten => s,1,Answer() exten => s,n,Dial(SIP/100) exten => s,n,Hangup exten => s,i,Hangup exten => 0033535541286,1,Dial(SIP/kalidev1) exten => 0033535541287,1,Dial(SIP/kalidev2) [appel-sortant] ; Cette partie gere les appels sortants exten => _X.,1,Dial(SIP/${EXTEN}@forfait-ovh) ; Sur cette ligne allons appeler en passant par la section [forfait-ovh] du fichier sip.conf [appel-sortant2] ; Cette partie gere les appels sortants exten => _X.,1,Dial(SIP/${EXTEN}@forfait-ovh2) ; Sur cette ligne allons appeler en passant par la section [forfait-ovh] du fichier sip.conf
J'ai des erreurs que je ne comprends pas 401 et 501 mon problème pourrait'il venir de là ?Code:<-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '490cc126558d31942b8acc4d67b75e01@127.0.1.1' in 32000 ms (Method: REGISTER) [Mar 21 15:25:22] NOTICE[7368]: chan_sip.c:18270 handle_response_register: Outbound Registration: Expiry for sip.ovh.net is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6968a22d;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as6bc0bd45 To: <sip:sip.ovh.net> Contact: <sip:asterisk@192.168.1.10> Call-ID: 49e3754a77bcf74b7d8280b527ced8f2@192.168.1.10 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.9-2 Date: Mon, 21 Mar 2011 14:25:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.17:5060 ---> SIP/2.0 501 Not Implemented Allow: UPDATE,REFER,INFO Call-ID: 49e3754a77bcf74b7d8280b527ced8f2@192.168.1.10 CSeq: 102 OPTIONS From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as6bc0bd45 Server: Cirpack/v4.42s (gw_sip) To: <sip:sip.ovh.net>;tag=00-07978-5df31608-2a73d8780 Via: SIP/2.0/UDP 192.168.1.10:5060;received=87.90.83.115;rport=5060;branch=z9hG4bK6968a22d Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '49e3754a77bcf74b7d8280b527ced8f2@192.168.1.10' Method: OPTIONS Reliably Transmitting (NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK49f2c98f;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as73a5ee28 To: <sip:sip.ovh.net> Contact: <sip:asterisk@192.168.1.10> Call-ID: 16b1e1145f9b677679e08e521b8a0fc4@192.168.1.10 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.9-2 Date: Mon, 21 Mar 2011 14:25:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.17:5060 ---> SIP/2.0 501 Not Implemented Allow: UPDATE,REFER,INFO Call-ID: 16b1e1145f9b677679e08e521b8a0fc4@192.168.1.10 CSeq: 102 OPTIONS From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as73a5ee28 Server: Cirpack/v4.42s (gw_sip) To: <sip:sip.ovh.net>;tag=00-08082-5df31615-1828b0b05 Via: SIP/2.0/UDP 192.168.1.10:5060;received=87.90.83.115;rport=5060;branch=z9hG4bK49f2c98f Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '16b1e1145f9b677679e08e521b8a0fc4@192.168.1.10' Method: OPTIONS <--- 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=z9hG4bK9eba6430d94b91cb6df745c95c5319f8;rport From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=250346009 To: "kalidev 2" <sip:kalidev2@192.168.1.10> Call-ID: 1219423529@192_168_1_20 CSeq: 12906 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 (NAT) to 192.168.1.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK9eba6430d94b91cb6df745c95c5319f8;received=192.168.1.20;rport=5060 From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=250346009 To: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=as027ba0fa Call-ID: 1219423529@192_168_1_20 CSeq: 12906 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="3a208742" Content-Length: 0
J'espère que vous aurez tous les éléments pour me sortir de se problème.
Merci de votre aide
Cordialement
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