Affichage des résultats 1 à 2 sur 2

Discussion: Timeout lors du ringing

  1. #1
    Membre Junior
    Date d'inscription
    mai 2013
    Messages
    3
    Downloads
    0
    Uploads
    0

    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 ) 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
    ---
    
    cvgVMasterisk1*CLI> 
    Scheduling 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

  2. #2
    Membre Junior
    Date d'inscription
    mai 2013
    Messages
    3
    Downloads
    0
    Uploads
    0
    J'ai finalement trouvé l'origine du problème avec l'aide du forum Asterisk US.

    En fait, je réalisais une commande AMI de type originate sur un local channel. Je redirigeais ensuite le local channel vers un context contenant un Dial. Comme je n'avais pas défini de timeout sur cette commande, j'avais un timeout sur le local channel au bout de 30 sec (paramétrage par défaut) qui entrainait un raccrochage sur le channel du dial.


    J'ai donc simplement ajouter le paramètre timeout à ma commande AMI et tout fonctionne.

Les tags pour cette discussion

Règles de messages

  • Vous ne pouvez pas créer de nouvelles discussions
  • Vous ne pouvez pas envoyer des réponses
  • Vous ne pouvez pas envoyer des pièces jointes
  • Vous ne pouvez pas modifier vos messages
  •