Timeout lors du ringing

Bonjour à tous,

Je rencontre un petit souci lors d'un Dial depuis un serveur asterisk vers un trunk SIP (type completel mais également cisco), la sonnerie s'arrête après environ 30 secondes, indépendamment du timeout spécifié. (Si le timeout est inférieur à 30 sec, la sonnerie s'arrête bien au temps désiré, mais ne dépasse jamais les 30 sec).

J'ai lu ici et là qu'il s'agissait d'un problème de NAT, mais je n'ai pas trouvé (ou compris :confused:) de solutions. Je passe par un proxy SIP Kamailio. J'utilise la version d'Asterisk 1.8.17.0 que j'ai installé classiquement depuis les sources.

J'ai fait un debug SIP :

Code:

INVITE sip:00582743866@10.1.9.5 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport Max-Forwards: 70 From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 To: <sip:00582743866@10.1.9.5> Contact: <sip:0123456789@10.1.3.233:5060> Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.17.0 Date: Tue, 28 May 2013 07:27:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 231 v=0 o=root 204944793 204944793 IN IP4 10.1.3.233 s=Asterisk PBX 1.8.17.0 c=IN IP4 10.1.3.233 t=0 0 m=audio 16486 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.1.3.240:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport=5060 From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 To: <sip:00582743866@10.1.9.5> Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 CSeq: 102 INVITE Server: kamailio (3.3.0 (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.1.3.240:5060 ---> SIP/2.0 183 Session Progress Date: Tue, 28 May 2013 07:32:22 GMT Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 Allow-Events: telephone-event Supported: sdp-anat Remote-Party-ID: <sip:00582743866@10.1.9.5>;party=called;screen=no;privacy=off Content-Length: 179 To: <sip:00582743866@10.1.9.5>;tag=D0C52F2C-1FEC Contact: <sip:00582743866@10.1.9.5:5060> C Content-Type: application/sdp Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport=5060 CSeq: 102 INVITE Server: Cisco-SIPGateway/IOS-12.x Record-Route: <sip:10.1.3.240;lr=on> v=0 o=CiscoSystemsSIP-GW-UserAgent 7232 6826 IN IP4 10.1.9.5 s=SIP Call c=IN IP4 10.1.9.5 t=0 0 m=audio 17098 RTP/AVP 0 c=IN IP4 10.1.9.5 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.1.9.5:17098 <--- SIP read from UDP:10.1.3.240:5060 ---> SIP/2.0 183 Session Progress Date: Tue, 28 May 2013 07:32:22 GMT Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 Allow-Events: telephone-event Supported: sdp-anat Remote-Party-ID: <sip:00582743866@10.1.9.5>;party=called;screen=no;privacy=off Content-Length: 179 To: <sip:00582743866@10.1.9.5>;tag=D0C52F2C-1FEC Contact: <sip:00582743866@10.1.9.5:5060> C Content-Type: application/sdp Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport=5060 CSeq: 102 INVITE Server: Cisco-SIPGateway/IOS-12.x Record-Route: <sip:10.1.3.240;lr=on> v=0 o=CiscoSystemsSIP-GW-UserAgent 7232 6826 IN IP4 10.1.9.5 s=SIP Call c=IN IP4 10.1.9.5 t=0 0 m=audio 17098 RTP/AVP 0 c=IN IP4 10.1.9.5 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> list_route: hop: <sip:10.1.3.240;lr=on> -- SIP/cisco-000000ce is making progress passing it to Local/s@dso-command-00002a7d;2 OPTIONS sip:10.1.3.240 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK3d7d9839;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.1.3.233>;tag=as55bed21d To: <sip:10.1.3.240> Contact: <sip:asterisk@10.1.3.233:5060> Call-ID: 75b4b24424b79ed77a61b9931ceafcde@10.1.3.233:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.17.0 Date: Tue, 28 May 2013 07:28:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.1.3.240:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK3d7d9839;rport=5060 From: "asterisk" <sip:asterisk@10.1.3.233>;tag=as55bed21d To: <sip:10.1.3.240>;tag=b27e1a1d33761e85846fc98f5f3a7e58.dd62 Call-ID: 75b4b24424b79ed77a61b9931ceafcde@10.1.3.233:5060 CSeq: 102 OPTIONS Server: kamailio (3.3.0 (x86_64/linux)) Content-Length: 0 <-------------> Really destroying SIP dialog '75b4b24424b79ed77a61b9931ceafcde@10.1.3.233:5060' Method: OPTIONS OPTIONS sip:10.1.9.5 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK4eb93c1b;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.1.3.233>;tag=as6e2ee137 To: <sip:10.1.9.5> Contact: <sip:asterisk@10.1.3.233:5060> Call-ID: 1796e674563be7cb40eaf7bb3123a7d2@10.1.3.233:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.17.0 Date: Tue, 28 May 2013 07:28:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Content-Length: 0 --- SIP/2.0 200 OK Date: Tue, 28 May 2013 07:32:46 GMT Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER From: "asterisk" <sip:asterisk@10.1.3.233>;tag=as6e2ee137 Allow-Events: telephone-event Supported: 100rel,resource-priority,replaces,sdp-anat Content-Length: 157 To: <sip:10.1.9.5>;tag=D0C58E60-1A1D Content-Type: application/sdp Call-ID: 1796e674563be7cb40eaf7bb3123a7d2@10.1.3.233:5060 Accept: application/sdp Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK4eb93c1b;rport Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS v=0 o=CiscoSystemsSIP-GW-UserAgent 5489 5914 IN IP4 10.1.9.5 s=SIP Call c=IN IP4 10.1.9.5 t=0 0 m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3 c=IN IP4 10.1.9.5 <-------------> --- (14 headers 7 lines) --- Really destroying SIP dialog '1796e674563be7cb40eaf7bb3123a7d2@10.1.3.233:5060' Method: OPTIONS CANCEL sip:00582743866@10.1.9.5 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport Max-Forwards: 70 From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 To: <sip:00582743866@10.1.9.5> Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.8.17.0 Content-Length: 0 <--- SIP read from UDP:10.1.3.240:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport=5060 From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 To: <sip:00582743866@10.1.9.5>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2a48 Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 CSeq: 102 CANCEL Server: kamailio (3.3.0 (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- P read from UDP:10.1.3.240:5060 ---> SIP/2.0 487 Request Cancelled Reason: Q.850;cause=16 Date: Tue, 28 May 2013 07:32:51 GMT From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 Allow-Events: telephone-event Content-Length: 0 To: <sip:00582743866@10.1.9.5>;tag=D0C52F2C-1FEC Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport=5060 CSeq: 102 INVITE Server: Cisco-SIPGateway/IOS-12.x <-------------> Transmitting (NAT) to 10.1.3.240:5060: ACK sip:00582743866@10.1.9.5:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.233:5060;branch=z9hG4bK1121c8b3;rport Max-Forwards: 70 From: "0123456789" <sip:0123456789@10.1.3.233>;tag=as087c91b7 To: <sip:00582743866@10.1.9.5>;tag=D0C52F2C-1FEC Contact: <sip:0123456789@10.1.3.233:5060> Call-ID: 793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.17.0 Content-Length: 0 --- [KcvgVMasterisk1*CLI> [0KScheduling destruction of SIP dialog '793ea1960b6db6961d64db3618655a7c@10.1.3.233:5060' in 12800 ms (Method: INVITE)

Et voici ma configuration sip.conf

Code:

[general] context=incoming-sip udpbindaddr=10.1.3.233 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes qualify=yes disallow=all allow=ulaw outboundproxy=10.1.3.240 t1min=200 timert1=700 timerb=45000 rtptimeout=60 rtpholdtimeout=300 rtpkeepalive=15 allowsubscribe=no directmedia=no [authentication] [basic-options](!) dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) nat=yes directmedia=no host=dynamic [public-phone](!,basic-options) nat=no directmedia=yes [my-codecs](!) disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw [ulaw-phone](!) disallow=all allow=ulaw #include </etc/asterisk/peers/*.conf>

Si quelqu'un a une piste, je suis preneur.

Merci d'avance,

Eric