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Discussion: Trunk SIP - Problème One way audio

  1. #1
    Membre Junior
    Date d'inscription
    juillet 2012
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    Unhappy Trunk SIP - Problème One way audio

    Bonjour à tous,

    J'essaie desespérement depuis plusieurs jours de faire fonctionner mon serveur Asterisk avec un trunk sip (il était auparavant branché avec un B410P et j'avais aussi galéré à l'époque pour trouver les bons paramètres). Il s'agit d'un trunk sip pris chez OVH

    Malheureusement j'ai un one way audio et je n'arrive pas à trouver la solution tout seul. Le problème est le suivant : lorsque quelqu'un m'appelle, je l'entends mais il ne m'entend pas

    Le paramétrage est le suivant :

    SIP.conf :

    Code:
    [general]
    nat=force_rport,comedia
    externip=XXX.XXX.XXX.XXX ; Adresse IP de la ligne SERANVILLE
    localnet=192.168.130.0/255.255.255.0
    qualify=yes
    defaultexpiry=1800 ; Temps de register de la ligne
    context=trunk-ovh ; Nom du context pour le trunk dans sip.conf
    directmedia=no
    bindport=5060 ; Port d'ecoute.
    bindaddr=0.0.0.0 
    srvlookup=no ;Autoriser les appels via noms DNS
    register => XXX@siptrunk.ovh.net ;Autenthification du trunk
    disallow=all
    allow=ulaw
    allow=alaw
    
    [trunk-ovh] 
    nat=force_rport,comedia
    type=friend
    host=siptrunk.ovh.net
    context=from-pstn
    language=fr
    insecure=invite,port
    defaultuser=XXX
    secret=XXX
    
    [215]
    type=friend
    defaultuser=215
    secret=XXX
    callerid="Audrey SER" <215>
    qualify=200        
    nat=force_rport,comedia       
    insecure=port  
    host=dynamic                  
    directmedia=no
    context=appel-sortant-comptoir
    language=fr
    call-limit=4
    busy-level=1
    subscribecontext=blf 
    callcounter=yes
    Firewall (tout est quasiment ouvert) => pour info j'ai laissé la config de base de rtp.conf. Client étant mon réseau interne 192.168.130.X

    Code:
    iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
    iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
    iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
    
    iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p udp --dport 5060 -m state --state NEW,ESTABLISHED -j ACCEPT
    iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p udp --sport 5060 -m state ! --state NEW -j ACCEPT
    iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p tcp --dport 5060 -m state --state NEW,ESTABLISHED -j ACCEPT
    iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p tcp --sport 5060 -m state ! --state NEW -j ACCEPT
    iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p udp --dport 5962 -m state --state NEW,ESTABLISHED -j ACCEPT
    iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p udp --sport 5962 -m state ! --state NEW -j ACCEPT
    iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
    Je poste le debug sip en dessous

    Merci d'avance pour votre aide

  2. #2
    Membre Junior
    Date d'inscription
    juillet 2012
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    Code:
    <--- SIP read from UDP:91.121.129.23:5060 --->
    INVITE sip:s@192.168.2.100:5060;transport=udp SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    Content-Type: application/sdp
    CSeq: 198813419 INVITE
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    To: <sip:0383720044@10.7.1.65;user=phone>
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f
    Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 318
    
    v=0
    o=cp10 156526349186 156526349186 IN IP4 91.121.128.136
    s=SIP Call
    c=IN IP4 91.121.128.136
    t=0 0
    m=audio 32142 RTP/AVP 8 18 0 101
    b=AS:82
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    <------------->
    --- (13 headers 15 lines) ---
    Sending to 91.121.129.23:5060 (NAT)
    Sending to 91.121.129.23:5060 (NAT)
    Using INVITE request as basis request - 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Found peer 'trunk-ovh' for '0383723596' from 91.121.129.23:5060
    Found RTP audio format 8
    Found RTP audio format 18
    Found RTP audio format 0
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format G729 for ID 18
    Found audio description format PCMU for ID 0
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 91.121.128.136:32142
    Looking for s in from-pstn (domain 192.168.2.100)
    sip_route_dump: route/path hop: <sip:91.121.129.23:5060;lr>
    
    <--- Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813419 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Length: 0
    
    
    <------------>
    
    <--- Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813419 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Length: 0
    
    
    <------------>
    Audio is at 11998
    Adding codec ulaw to SDP
    Adding codec alaw to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 192.168.130.49:5060:
    INVITE sip:214@192.168.130.49:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport
    Max-Forwards: 70
    From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    To: <sip:214@192.168.130.49:5060>
    Contact: <sip:0383723596@192.168.130.254:5060>
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Date: Thu, 08 Aug 2019 11:24:51 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 294
    
    v=0
    o=root 638829780 638829780 IN IP4 192.168.130.254
    s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
    c=IN IP4 192.168.130.254
    t=0 0
    m=audio 11998 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=maxptime:150
    a=sendrecv
    
    ---
    
    <--- SIP read from UDP:192.168.130.49:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
    From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    To: <sip:214@192.168.130.49:5060>
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 102 INVITE
    User-Agent: Yealink SIP-T41S 66.84.0.15
    Content-Length: 0
    
    <------------->
    --- (8 headers 0 lines) ---
    
    <--- SIP read from UDP:192.168.130.49:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
    From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    To: <sip:214@192.168.130.49:5060>;tag=2669272485
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 102 INVITE
    Contact: <sip:214@192.168.130.49:5060>
    Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
    User-Agent: Yealink SIP-T41S 66.84.0.15
    Allow-Events: talk,hold,conference,refer,check-sync
    Content-Length: 0
    
    <------------->
    --- (11 headers 0 lines) ---
    sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>
    
    <--- Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813419 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Length: 0
    
    
    <------------>
    
    <--- SIP read from UDP:192.168.130.49:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
    From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    To: <sip:214@192.168.130.49:5060>;tag=2669272485
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 102 INVITE
    Contact: <sip:214@192.168.130.49:5060>
    Content-Type: application/sdp
    Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
    User-Agent: Yealink SIP-T41S 66.84.0.15
    Allow-Events: talk,hold,conference,refer,check-sync
    Content-Length: 215
    
    v=0
    o=- 20037 20037 IN IP4 192.168.130.49
    s=SDP data
    c=IN IP4 192.168.130.49
    t=0 0
    m=audio 12244 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    <------------->
    --- (12 headers 11 lines) ---
    Found RTP audio format 0
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.130.49:12244
    sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>
    Transmitting (NAT) to 192.168.130.49:5060:
    ACK sip:214@192.168.130.49:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK1c2f34b1;rport
    Max-Forwards: 70
    From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    To: <sip:214@192.168.130.49:5060>;tag=2669272485
    Contact: <sip:0383723596@192.168.130.254:5060>
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Content-Length: 0

  3. #3
    Membre Junior
    Date d'inscription
    juillet 2012
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    Code:
    ---
    Audio is at 10246
    Adding codec ulaw to SDP
    Adding codec alaw to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    
    <--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813419 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Type: application/sdp
    Content-Length: 292
    
    v=0
    o=root 665561663 665561663 IN IP4 185.246.18.202
    s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
    c=IN IP4 185.246.18.202
    t=0 0
    m=audio 10246 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=maxptime:150
    a=sendrecv
    
    <------------>
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    ACK sip:s@192.168.2.100:5060 SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    CSeq: 198813419 ACK
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-LHLE-8707918f-3db175f4
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 0
    
    <------------->
    --- (10 headers 0 lines) ---
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    INVITE sip:s@192.168.2.100:5060 SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    Content-Type: application/sdp
    CSeq: 198813420 INVITE
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6
    Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 318
    
    v=0
    o=cp10 156526349186 156526349187 IN IP4 91.121.128.136
    s=SIP Call
    c=IN IP4 91.121.128.136
    t=0 0
    m=audio 32142 RTP/AVP 0 8 18 101
    b=AS:82
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    <------------->
    --- (13 headers 15 lines) ---
    Sending to 91.121.129.23:5060 (NAT)
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 18
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Found audio description format G729 for ID 18
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 91.121.128.136:32142
    
    <--- Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813420 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Length: 0
    
    
    <------------>
    Audio is at 10246
    Adding codec ulaw to SDP
    Adding codec alaw to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    
    <--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813420 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Type: application/sdp
    Content-Length: 292
    
    v=0
    o=root 665561663 665561664 IN IP4 185.246.18.202
    s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
    c=IN IP4 185.246.18.202
    t=0 0
    m=audio 10246 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=maxptime:150
    a=sendrecv
    
    <------------>
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    ACK sip:s@192.168.2.100:5060 SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    CSeq: 198813420 ACK
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-YYHU-87079197-2495cec1
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 0
    
    <------------->
    --- (10 headers 0 lines) ---
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    INVITE sip:s@192.168.2.100:5060 SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    Content-Type: application/sdp
    CSeq: 198813421 INVITE
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb
    Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 243
    
    v=0
    o=cp10 156526349186 156526349188 IN IP4 91.121.128.136
    s=SIP Call
    c=IN IP4 91.121.128.136
    t=0 0
    m=audio 32142 RTP/AVP 8 101
    b=AS:82
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    <------------->
    --- (13 headers 12 lines) ---
    Sending to 91.121.129.23:5060 (NAT)
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 91.121.128.136:32142
    
    <--- Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813421 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Length: 0

  4. #4
    Membre Junior
    Date d'inscription
    juillet 2012
    Messages
    14
    Downloads
    0
    Uploads
    0
    Code:
    <------------>
    Audio is at 10246
    Adding codec alaw to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    
    <--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
    Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 198813421 INVITE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:s@185.246.18.202:5060>
    Content-Type: application/sdp
    Content-Length: 268
    
    v=0
    o=root 665561663 665561665 IN IP4 185.246.18.202
    s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
    c=IN IP4 185.246.18.202
    t=0 0
    m=audio 10246 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:150
    a=sendrecv
    
    <------------>
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    ACK sip:s@192.168.2.100:5060 SIP/2.0
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    Contact: <sip:10.7.1.65:5060>
    CSeq: 198813421 ACK
    From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Max-Forwards: 29
    To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-SAQW-8707921f-562ea85e
    User-Agent: Cirpack/v4.76 (gw_sip)
    Content-Length: 0
    
    <------------->
    --- (10 headers 0 lines) ---
    
    <--- SIP read from UDP:192.168.130.49:5060 --->
    BYE sip:0383723596@192.168.130.254:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565
    From: <sip:214@192.168.130.49:5060>;tag=2669272485
    To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 2 BYE
    Contact: <sip:214@192.168.130.49:5060>
    Max-Forwards: 70
    User-Agent: Yealink SIP-T41S 66.84.0.15
    Content-Length: 0
    
    <------------->
    --- (10 headers 0 lines) ---
    Sending to 192.168.130.49:5060 (NAT)
    Scheduling destruction of SIP dialog '56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060' in 6400 ms (Method: BYE)
    
    <--- Transmitting (NAT) to 192.168.130.49:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565;received=192.168.130.49;rport=5060
    From: <sip:214@192.168.130.49:5060>;tag=2669272485
    To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
    Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
    CSeq: 2 BYE
    Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' in 6400 ms (Method: ACK)
    Reliably Transmitting (NAT) to 91.121.129.23:5060:
    BYE sip:10.7.1.65:5060 SIP/2.0
    Via: SIP/2.0/UDP 185.246.18.202:5060;branch=z9hG4bK26126eb9;rport
    Route: <sip:91.121.129.23:5060;lr>
    Max-Forwards: 70
    From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 102 BYE
    User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0
    
    
    ---
    
    <--- SIP read from UDP:91.121.129.23:5060 --->
    SIP/2.0 200 OK
    Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
    CSeq: 102 BYE
    From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
    Record-Route: <sip:91.121.129.23:5060;transport=udp;lr>;session=443934
    To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
    Via: SIP/2.0/UDP 192.168.2.100:5060;received=192.168.2.100;rport=5060;branch=z9hG4bK26126eb9
    Server: Cirpack/v4.76 (gw_sip)
    Content-Length: 0
    
    <------------->
    --- (9 headers 0 lines) ---
    SIP Response message for INCOMING dialog BYE arrived
    Really destroying SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' Method: ACK

  5. #5
    Membre Junior
    Date d'inscription
    aot 2016
    Messages
    13
    Downloads
    0
    Uploads
    0
    Bonjour,

    J'ai déjà rencontré ce problème plusieurs, c'est généralement toujours un problème "réseau" dans 99% des cas, une fois c’était effectivement un blocage niveau firewall et l'autre cas que j'ai rencontré était un peu plus "tordu", c’était une route de retour qui n'empruntait pas le même chemin que la requête initiale.

    Cordialement,

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