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Asterisk beendet SIP-Verbindung nicht

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von Fuso, 21 Nov. 2008.

  1. Fuso

    Fuso Neuer User

    Registriert seit:
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    Hallo,

    ich habe das Problem das Asterisk (SVN-branch-1.4-r157503) eine Festnetzverbindung nicht trennt wenn der interne SIP Teilnehmer (SNOM 320) auflegt. Als Anbieter habe ich 1&1. Raus und rein telefonieren klappt einwandfrei. Wenn der externe Anrufer auflegt wird alles ordnungsgemäß getrennt. Aus irgendeinem Grund versteht 1&1 das BYE Kommando nicht.

    Hier meine Einstellungsdateien:

    sip.conf:
    Code:
    [general]
    context=default
    bindport=5060
    bindaddr=0.0.0.0
    srvlookup=yes
    language=de
    externip=xxx.dyndns.org
    localnet=192.168.110.0/255.255.255.0
    qualify=no
    disallow=all
    allow=alaw
    allow=ulaw
    allow=g729
    allow=gsm
    allow=slinear
    nat=no
    allowsubscribe = yes
    notifyringing = yes
    notifyhold = yes
    limitonpeers = yes
    externrefresh = 15
    
    register => 4933xxxxxxxx:xxxxxxxx@sip.1und1.de/4933xxxxxxxx
    
    [4933xxxxxxxx]
    type=friend
    username=4933xxxxxxxx
    fromuser=4933xxxxxxxx
    secret=xxxxxxxx
    host=sip.1und1.de
    fromdomain=sip.1und1.de
    nat=yes
    insecure=port,invite
    caninvite=no
    canreinvite=no
    
    1und1_in_0]
    type=friend
    fromdomain=sipbalance0.1und1.de
    host=sipbalance0.1und1.de
    insecure=port,invite
    nat=yes
    context=ankommend
    caninvite=no
    canreinvite=no
    
    [1und1_in_1]
    type=friend
    fromdomain=sipbalance1.1und1.de
    host=sipbalance1.1und1.de
    insecure=port,invite
    nat=yes
    context=ankommend
    caninvite=no
    canreinvite=no
    
    [10]
    callerid=Privat <10>
    host=dynamic
    domain=192.168.110.124
    user=10
    secret=1313
    type=friend
    mailbox=10
    nat=no
    caninvite=no
    canreinvite=no
    context=default
    subscribecontext=default
    call-limit = 10
    callgroup = 2
    pickupgroup = 2
    
    [11]
    callerid=Büro <11>
    host=dynamic
    domain=192.168.110.124
    user=11
    secret=7990
    type=friend
    mailbox=11
    nat=no
    caninvite=no
    canreinvite=no
    context=default
    subscribecontext=default
    call-limit = 10
    callgroup = 2
    pickupgroup = 2
    
    [12]
    callerid=Büro Handy <12>
    host=dynamic
    domain=192.168.110.124
    user=12
    secret=7990
    type=friend
    mailbox=11
    vmexten=11
    nat=no
    caninvite=no
    canreinvite=no
    context=default
    subscribecontext=default
    call-limit = 10
    callgroup = 2
    pickupgroup = 2
    
    extensions.conf:
    Code:
    [general]
    static=yes
    writeprotect=no
    
    [echotest]
    exten => 81,1,answer
    exten => 81,2,wait(1)
    exten => 81,3,playback(demo-echotest)
    exten => 81,4,echo
    exten => 81,5,playback(demo-echodone)
    exten => 81,6,hangup
    
    [mailbox]
    exten => 80,1,answer
    exten => 80,n,wait(1)
    exten => 80,n,voicemailmain
    exten => 80,n,hangup
    
    [mailbox_own]
    exten => 88,1,answer
    exten => 88,n,wait(1)
    exten => 88,n,voicemailmain(s${CALLERID(num)})
    exten => 88,n,hangup
    
    exten => asterisk,1,VoicemailMain(s${CALLERID(num)})
    
    [hear_music]
    exten => 99,1,answer
    exten => 99,n,wait(1)
    exten => 99,n,musiconhold(mp3)
    
    [lokal]
    exten => 10,hint,SIP/10
    exten => 10,1,Answer
    exten => 10,n,Dial(SIP/10,55,Ttrm)
    exten => 10,n,Voicemail(10,u)
    exten => 11,n,Hangup
    
    exten => 11,hint,SIP/11
    exten => 11,1,Answer
    exten => 11,n,Dial(SIP/11,55,Ttrm)
    exten => 11,n,Voicemail(11,u)
    exten => 11,n,Hangup
    
    exten => 12,hint,SIP/12
    exten => 12,1,Answer
    exten => 12,n,Dial(SIP/12,55,Ttrm)
    exten => 12,n,Voicemail(12,u)
    exten => 12,n,Hangup
    
    [1und1_out]
    exten => _0.,1,Dial(SIP/${EXTEN:1}@4933xxxxxxxx,45,r)
    
    [ankommend]
    exten => 4933xxxxxxxx,1,NoOp(Anruf auf 1und1)
    exten => 4933xxxxxxxx,n,Ringing
    exten => 4933xxxxxxxx,n,Set(CALLERID(all)=1und1: ${CALLERID(num)} <${CALLERID(num)}>)
    exten => 4933xxxxxxxx,n,Dial(SIP/11,30,t)
    exten => 4933xxxxxxxx,n,Goto(r-${DIALSTATUS},1)
    
    exten => r-CONGESTION,1,voicemail(11,u)
    exten => r-CONGESTION,2,Hangup
    
    exten => r-BUSY,1,voicemail(11,b)
    exten => r-BUSY,2,Hangup
    
    exten => r-NOANSWER,1,voicemail(11,u)
    exten => r-NOANSWER,2,Hangup
    
    exten => r-CHANUNAVAIL,1,voicemail(11,u)
    exten => r-CHANUNAVAIL,2,Hangup
    
    [default]
    include => lokal
    include => echotest
    include => hear_music
    include => mailbox
    include => mailbox_own
    include => 1und1_out
    
    Log von CLI:
    Code:
    <--- Reliably Transmitting (NAT) to 212.227.15.231:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 212.227.15.231;branch=z9hG4bK85a9.e531208e7daea31d3d1b0b327c2918c1.0;received=212.227.15.231
    Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK85a9.e531208e7daea31d3d1b0b327c2918c1.0
    Via: SIP/2.0/UDP 212.227.15.231;branch=z9hG4bK85a9.aac91692.0
    Via: SIP/2.0/UDP 195.71.47.146;branch=z9hG4bK85a9.aac91692.0
    Via: SIP/2.0/UDP 1und1-8.sip.mgc.voip.telefonica.de:5060;received=193.189.245.140;branch=z9hG4bKterm-9d4279-+4933xxxxxxxx-+4933caller-18517
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.232;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:195.71.47.146;lr=on;ftag=132354529>
    From: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    To: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 1 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:4933xxxxxxxx@79.192.170.244>
    Content-Type: application/sdp
    Content-Length: 335
    
    v=0
    o=root 16489 16489 IN IP4 79.192.170.244
    s=session
    c=IN IP4 79.192.170.244
    t=0 0
    m=audio 19214 RTP/AVP 8 0 18 3 99
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    
    <------------>
    
    <--- SIP read from 212.227.15.231:5060 --->
    ACK sip:4933xxxxxxxx@192.168.110.124:5060 SIP/2.0
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.232;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:195.71.47.146;lr=on;ftag=132354529>
    Via: SIP/2.0/UDP 212.227.15.231:5060;branch=z9hG4bK85a9.e531208e7daea31d3d1b0b327c2918c1.2
    Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK85a9.e531208e7daea31d3d1b0b327c2918c1.2
    Via: SIP/2.0/UDP 212.227.15.231;branch=z9hG4bK85a9.aac91692.2
    Via: SIP/2.0/UDP 195.71.47.146;branch=z9hG4bK85a9.aac91692.2
    Via: SIP/2.0/UDP 1und1-8.sip.mgc.voip.telefonica.de:5060;received=193.189.245.140;branch=z9hG4bKterm-9d4279-+4933xxxxxxxx-+4933caller-26028
    From: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    To: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 1 ACK
    Max-Forwards: 15
    Content-Length: 0
    P-hint: rr-enforced
    
    
    <------------->
    --- (17 headers 0 lines) ---
    Really destroying SIP dialog '182f8a3e0aea74656549c1bb216ea036@127.0.0.2' Method: REGISTER
    
    <--- SIP read from 192.168.110.60:2163 --->
    BYE sip:+4933caller@192.168.110.124 SIP/2.0
    Via: SIP/2.0/UDP 192.168.110.60:2163;branch=z9hG4bK-xhto2zbi6rzw;rport
    From: <sip:11@192.168.110.60:2163;line=sjml8wz8>;tag=kuxokby3l2
    To: "1und1: +4933caller" <sip:+4933caller@192.168.110.124>;tag=as333934b2
    Call-ID: 06e7d8b37a9230aa22f6586b2cb35420@192.168.110.124
    CSeq: 1 BYE
    Max-Forwards: 70
    Contact: <sip:11@192.168.110.60:2163;line=sjml8wz8>;flow-id=1
    User-Agent: snom320/7.1.17
    RTP-RxStat: Total_Rx_Pkts=234,Rx_Pkts=234,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
    RTP-TxStat: Total_Tx_Pkts=248,Tx_Pkts=248,Remote_Tx_Pkts=0
    Content-Length: 0
    
    
    <------------->
    --- (12 headers 0 lines) ---
    Sending to 192.168.110.60 : 2163 (NAT)
    
    <--- Transmitting (NAT) to 192.168.110.60:2163 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.110.60:2163;branch=z9hG4bK-xhto2zbi6rzw;received=192.168.110.60;rport=2163
    From: <sip:11@192.168.110.60:2163;line=sjml8wz8>;tag=kuxokby3l2
    To: "1und1: +4933caller" <sip:+4933caller@192.168.110.124>;tag=as333934b2
    Call-ID: 06e7d8b37a9230aa22f6586b2cb35420@192.168.110.124
    CSeq: 1 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:+4933caller@192.168.110.124>
    Content-Length: 0
    
    
    <------------>
      == Spawn extension (ankommend, 4933xxxxxxxx, 4) exited non-zero on 'SIP/5060-081e0d08'
    Scheduling destruction of SIP dialog 'e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de' in 32000 ms (Method: ACK)
    set_destination: Parsing <sip:212.227.15.231;lr=on;ftag=132354529> for address/port to send to
    set_destination: set destination to 212.227.15.231, port 5060
    Reliably Transmitting (NAT) to 212.227.15.231:5060:
    BYE sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060 SIP/2.0
    Via: SIP/2.0/UDP 79.192.170.244:5060;branch=z9hG4bK7fda52bb;rport
    Route: <sip:212.227.15.231;lr=on;ftag=132354529>,<sip:212.227.15.232;lr=on;ftag=132354529>,<sip:212.227.15.231;lr=on;ftag=132354529>,<sip:195.71.47.146;lr=on;ftag=132354529>
    From: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    To: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 102 BYE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    Really destroying SIP dialog '06e7d8b37a9230aa22f6586b2cb35420@192.168.110.124' Method: BYE
    
    <--- SIP read from 212.227.15.231:5060 --->
    SIP/2.0 400 Fehlerhafte SIP-Nachricht
    Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK7fda52bb;rport=5060
    From: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    To: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 102 BYE
    Server: UI OpenSER
    Content-Length: 0
    
    
    <------------->
    --- (8 headers 0 lines) ---
    SIP Response message for INCOMING dialog BYE arrived
        -- Incoming call: Got SIP response 400 "Fehlerhafte SIP-Nachricht" back from 212.227.15.231
    
    Rest des Logs von CLI nachdem ich den Anrufer aufgelegt habe:
    Code:
    <--- SIP read from 212.227.15.231:5060 --->
    BYE sip:4933xxxxxxxx@192.168.110.124:5060 SIP/2.0
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.232;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:195.71.47.146;lr=on;ftag=132354529>
    Via: SIP/2.0/UDP 212.227.15.231:5060;branch=z9hG4bK55a9.bbdc0d2ce51099cde091827fa54131dd.0
    Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK55a9.bbdc0d2ce51099cde091827fa54131dd.0
    Via: SIP/2.0/UDP 212.227.15.231;branch=z9hG4bK55a9.d5dca2e1.0
    Via: SIP/2.0/UDP 195.71.47.146;branch=z9hG4bK55a9.d5dca2e1.0
    Via: SIP/2.0/UDP 1und1-8.sip.mgc.voip.telefonica.de:5060;received=193.189.245.140;branch=z9hG4bKterm-9d4279-+4933xxxxxxxx-+4933caller-9890
    From: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    To: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 2 BYE
    Max-Forwards: 66
    Supported: timer
    Content-Length: 0
    P-hint: rr-enforced
    
    
    <------------->
    --- (18 headers 0 lines) ---
    Sending to 212.227.15.231 : 5060 (NAT)
    Scheduling destruction of SIP dialog 'e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de' in 32000 ms (Method: BYE)
    
    <--- Transmitting (NAT) to 212.227.15.231:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 212.227.15.231:5060;branch=z9hG4bK55a9.bbdc0d2ce51099cde091827fa54131dd.0;received=212.227.15.231
    Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK55a9.bbdc0d2ce51099cde091827fa54131dd.0
    Via: SIP/2.0/UDP 212.227.15.231;branch=z9hG4bK55a9.d5dca2e1.0
    Via: SIP/2.0/UDP 195.71.47.146;branch=z9hG4bK55a9.d5dca2e1.0
    Via: SIP/2.0/UDP 1und1-8.sip.mgc.voip.telefonica.de:5060;received=193.189.245.140;branch=z9hG4bKterm-9d4279-+4933xxxxxxxx-+4933caller-9890
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.232;lr=on;ftag=132354529>
    Record-Route: <sip:212.227.15.231;lr=on;ftag=132354529>
    Record-Route: <sip:195.71.47.146;lr=on;ftag=132354529>
    From: +4933caller <sip:+4933caller@1und1-8.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=132354529
    To: +4933xxxxxxxx <sip:+4933xxxxxxxx@195.71.47.146:5060;user=phone>;tag=as74c859b8
    Call-ID: e4c51b6-5c46ece4-465658dd-465a@subscriber5.interconnect.mgc.voip.telefonica.de
    CSeq: 2 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:4933xxxxxxxx@79.192.170.244>
    Content-Length: 0
    
    Ich hoffe das mir jemand erklären kann wo das Problem liegt.

    Vielen Dank
    Fuso
     
  2. laureen

    laureen Mitglied

    Registriert seit:
    17 Okt. 2004
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    ändere mal
    Code:
    ...
    externip=xxx.dyndns.org
    ...
    in
    Code:
    ...
    externhost=xxx.dyndns.org
    externrefresh=10
    ...
    denke mal, "externip" heisst so, weil dort eine ip adresse rein soll...bei solchen fehlern ist meistens nat/externe ip schuld...

    grüße,
    laureen
     
  3. Fuso

    Fuso Neuer User

    Registriert seit:
    20 Nov. 2008
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    Hallo Laureen,

    vielen Dank für deine Antwort. Du hast recht. Es lag am Natting. Habe mehr Ports mit DNAT weiter geleitet als gut war. Hab jetzt nur noch die RTP Ports drin und schon funktioniert es reibungslos.

    Gruß
    Fuso