.titleBar { margin-bottom: 5px!important; }

SIP / ENUM

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von jstocker, 29 Apr. 2005.

  1. jstocker

    jstocker Neuer User

    Registriert seit:
    2 März 2005
    Beiträge:
    4
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Irgendwie schaff ich es nicht einen annonymen SIP call in die weite Welt zu machen. Die Fehlermeldungen sehen wie folgt aus.

    Code:
      -- Executing EnumLookup("SIP/p200-b0e6", "493039833444") in new stack
    ENUM got '1'
        -- Executing Dial("SIP/p200-b0e6", "SIP/sf@snom.com|30") in new stack
        -- parse_srv: SRV mapped to host sip.snom.com, port 5082
        -- Called [email]sf@snom.com[/email]
    Apr 29 23:09:18 NOTICE[1503]: chan_sip.c:6880 handle_response: Failed to authenticate on INVITE to '"P200@Asterisk" <sip:Unknown@84.137.232.146>;tag=as4ec63708'
      == Spawn extension (sip-in-local, **493039833444, 2) exited non-zero on 'SIP/p200-b0e6'
        -- Got SIP response 481 "Call/Transaction does not exist" back from 217.115.141.99
    
    Der Name "P200@Asterisk" kommt direkt vom IP Phone und hat nix mit Asterisk selber zu tun. Die IP 84.137.232.146 ist zur Zeit die tatsächliche IP des Asterisk Servers und von draußen durch 5060 erreichbar.

    Extensions.conf
    Code:
    exten => _**.,1,EnumLookup(${EXTEN:2})
    exten => _**.,2,Dial(${ENUM},30)
    exten => _**.,3,Congestion
    
    Die ENUM Auflösung klappt einwandfrei, wie man an obiger Ausgabe sehen kann. Ich kann nun gar nichts mit der Fehlermeldung anfangen. Die Warnung stellt für mich auch Rätsel dar: Wieso will da sich das Telefon authentifizieren und wie soll das gehen, denn die beiden kennen sich ja nicht, das ist ja auch der Sinn eines ENUM Anrufes.

    Kann mir jemand da weiter helfen?

    Ich habe folgende Konfiguration

    Internet --> FW + Asterisk -> 10.x.x.x -> Hardphone

    Da Asterisk ebenfalls auf beiden Netzen horcht stellt sich nun die Frage des NAT. Wenn ich einen SIP call zu Asterisk mache (ueber 10.x.x.x) dieser dann einen Outcall ins Internet macht fungiert er hier ja hoffentlich als Gateway und zwingt das SIP Phone nicht die Verbindung selber aufzubauen (ich gehe jedenfalls ganz stark davon aus). Also sollte ich hier kein NAT Problem haben.
     
  2. Maik

    Maik Gesperrt

    Registriert seit:
    1 Apr. 2004
    Beiträge:
    1,778
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Ein 'sip debug' waere hier sehr hilfreich. Ausserdem waere es gut zu wissen, welche Asterisk-Version du verwendest.
     
  3. jstocker

    jstocker Neuer User

    Registriert seit:
    2 März 2005
    Beiträge:
    4
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Here you go...

    Asterisk 1.0.7


    Code:
    Sip read: 
    INVITE sip:**493039833444@10.1.1.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
    Session-Expires: 120
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307913 INVITE
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO
    Supported: timer,replaces
    User-Agent: P200 02.09
    Content-Type: application/sdp
    Content-Length: 224
    
    v=0
    o=p200 471593064 471593064 IN IP4 10.1.1.3
    s=P200 02.09
    c=IN IP4 10.1.1.3
    t=0 0
    m=audio 8000 RTP/AVP 0 18 4
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000/1
    a=sendrecv
    
    14 headers, 11 lines
    Using latest request as basis request
    Sending to 10.1.1.3 : 5060 (non-NAT)
    Reliably Transmitting (no NAT):
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as4f40f49e
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307913 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:**493039833444@10.1.1.1>
    Proxy-Authenticate: Digest realm="asterisk", nonce="317bd9b8"
    Content-Length: 0
    
    
     to 10.1.1.3:5060
    Scheduling destruction of call 'CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3' in 15000 ms
    Found user 'p200'
    
    
    Sip read: 
    ACK sip:**493039833444@10.1.1.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as4f40f49e
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307913 ACK
    User-Agent: P200 02.09
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Content-Length: 0
    
    
    10 headers, 0 lines
    
    
    Sip read: 
    INVITE sip:**493039833444@10.1.1.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    Session-Expires: 120
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 INVITE
    Proxy-Authorization: Digest username="p200", realm="asterisk", nonce="317bd9b8", opaque="", uri="sip:**493039833444@10.1.1.1:5060", response="f782a310c1849f462bfb763b55e90400"
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO
    Supported: timer,replaces
    User-Agent: P200 02.09
    Content-Type: application/sdp
    Content-Length: 224
    
    v=0
    o=p200 471593064 471593064 IN IP4 10.1.1.3
    s=P200 02.09
    c=IN IP4 10.1.1.3
    t=0 0
    m=audio 8000 RTP/AVP 0 18 4
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:18 G729/8000/1
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000/1
    a=sendrecv
    
    15 headers, 11 lines
    Using latest request as basis request
    Sending to 10.1.1.3 : 5060 (non-NAT)
    Found user 'p200'
    Found RTP audio format 0
    Found RTP audio format 18
    Found RTP audio format 4
    Peer audio RTP is at port 10.1.1.3:8000
    Found description format PCMU
    Found description format G729
    Found description format G723
    Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Looking for **493039833444 in sip-in-local
    list_route: hop: <sip:p200@10.1.1.3:5060>
    Transmitting (no NAT):
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as3a12d869
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:**493039833444@10.1.1.1>
    Content-Length: 0
     to 10.1.1.3:5060
        -- Executing EnumLookup("SIP/p200-f348", "493039833444") in new stack
    Urgent handler
    Urgent handler
    ENUM got '1'
        -- Executing Dial("SIP/p200-f348", "SIP/sf@snom.com|30") in new stack
        -- parse_srv: SRV mapped to host sip.snom.com, port 5082
    We're at 84.137.243.197 port 19316
    Answering/Requesting with root capability 0x4 (ulaw)
    Answering with capability 0x2 (gsm)
    Answering with capability 0x8 (alaw)
    Answering with non-codec capability 0x1 (telephone-event)
    12 headers, 12 lines
    Reliably Transmitting:
    INVITE sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Sat, 30 Apr 2005 09:54:42 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 263
    
    v=0
    o=root 731 731 IN IP4 84.137.243.197
    s=session
    c=IN IP4 84.137.243.197
    t=0 0
    m=audio 19316 RTP/AVP 0 3 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
     (no NAT) to 217.115.141.99:5082
        -- Called [email]sf@snom.com[/email]
    Urgent handler
    
    
    Sip read: 
    SIP/2.0 407 Proxy Authorization Required
    v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
    f: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    t: <sip:sf@snom.com:5082>;tag=wkwisk5kxi
    i: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest realm="default", nonce="994dd6e729dac4e077a227a7e77bb394", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    
    
    8 headers, 0 lines
    Transmitting:
    ACK sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>;tag=wkwisk5kxi
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Content-Length: 0
    
     (no NAT) to 217.115.141.99:5082
    We're at 84.137.243.197 port 19316
    Answering/Requesting with root capability 0x4 (ulaw)
    Answering with capability 0x2 (gsm)
    Answering with capability 0x8 (alaw)
    Answering with non-codec capability 0x1 (telephone-event)
    Reliably Transmitting:
    INVITE sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:sf@snom.com:5082", nonce="994dd6e729dac4e077a227a7e77bb394", response="afa4b70d50f3fe1b3c9068bfa8a4d48c", opaque=""
    Date: Sat, 30 Apr 2005 09:54:42 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 263
    
    v=0
    o=root 731 732 IN IP4 84.137.243.197
    s=session
    c=IN IP4 84.137.243.197
    t=0 0
    m=audio 19316 RTP/AVP 0 3 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
     (no NAT) to 217.115.141.99:5082
    Sip read: 
    SIP/2.0 407 Proxy Authorization Required
    v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
    f: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    t: <sip:sf@snom.com:5082>;tag=rzmlmd0t1d
    i: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 103 INVITE
    Proxy-Authenticate: Digest realm="default", nonce="e6700a020a0003e6e46abd1a3ff398d7", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    
    
    8 headers, 0 lines
    Transmitting:
    ACK sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>;tag=rzmlmd0t1d
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 103 ACK
    User-Agent: Asterisk PBX
    Content-Length: 0
    
     (no NAT) to 217.115.141.99:5082
    We're at 84.137.243.197 port 19316
    Answering/Requesting with root capability 0x4 (ulaw)
    Answering with capability 0x2 (gsm)
    Answering with capability 0x8 (alaw)
    Answering with non-codec capability 0x1 (telephone-event)
    Reliably Transmitting:
    INVITE sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 104 INVITE
    User-Agent: Asterisk PBX
    Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:sf@snom.com:5082", nonce="e6700a020a0003e6e46abd1a3ff398d7", response="954d22ec0c5ee679182fdf6b0704bc9f", opaque=""
    Date: Sat, 30 Apr 2005 09:54:42 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 263
    
    v=0
    o=root 731 733 IN IP4 84.137.243.197
    s=session
    c=IN IP4 84.137.243.197
    t=0 0
    
    m=audio 19316 RTP/AVP 0 3 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
     (no NAT) to 217.115.141.99:5082
    
    
    Sip read: 
    SIP/2.0 407 Proxy Authorization Required
    v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
    f: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    t: <sip:sf@snom.com:5082>;tag=64fvi35m9h
    i: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 104 INVITE
    Proxy-Authenticate: Digest realm="default", nonce="e15d731ddcbf6ee617eeed25c1ddfeee", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    
    
    8 headers, 0 lines
    Transmitting:
    ACK sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>;tag=64fvi35m9h
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 104 ACK
    User-Agent: Asterisk PBX
    Content-Length: 0
    
     (no NAT) to 217.115.141.99:5082
    Apr 30 11:54:42 NOTICE[731]: chan_sip.c:6880 handle_response: Failed to authenticate on INVITE to '"P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784'
    Urgent handler
    Urgent handler
    
    
    Sip read: 
    CANCEL sip:**493039833444@10.1.1.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 CANCEL
    User-Agent: P200 02.09
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Content-Length: 0
    
    
    10 headers, 0 lines
    Sending to 10.1.1.3 : 5060 (non-NAT)
    Reliably Transmitting (no NAT):
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as3a12d869
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:**493039833444@10.1.1.1>
    Content-Length: 0
    
    
     to 10.1.1.3:5060
    Transmitting (no NAT):
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as3a12d869
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 CANCEL
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:**493039833444@10.1.1.1>
    Content-Length: 0
    
    
     to 10.1.1.3:5060
    Reliably Transmitting:
    CANCEL sip:sf@snom.com:5082 SIP/2.0
    Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
    From: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    To: <sip:sf@snom.com:5082>
    Contact: <sip:Unknown@84.137.243.197>
    Call-ID: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 104 CANCEL
    User-Agent: Asterisk PBX
    Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:sf@snom.com:5082", nonce="e6700a020a0003e6e46abd1a3ff398d7", response="3224127ffd897fdae81f25788c5c904c", opaque=""
    Content-Length: 0
    
     (no NAT) to 217.115.141.99:5082
    Scheduling destruction of call '73fa4da836008c145dec90be5612da1e@84.137.243.197' in 15000 ms
    
    
    Sip read: 
    SIP/2.0 481 Call/Transaction does not exist
    v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
    f: "P200@Asterisk" <sip:Unknown@84.137.243.197>;tag=as062fb784
    
    t: <sip:sf@snom.com:5082>
    i: 73fa4da836008c145dec90be5612da1e@84.137.243.197
    CSeq: 104 CANCEL
    l: 0
    
    
    7 headers, 0 lines
        -- Got SIP response 481 "Call/Transaction does not exist" back from 217.115.141.99
    Destroying call '73fa4da836008c145dec90be5612da1e@84.137.243.197'
    
    
    Sip read: 
    ACK sip:**493039833444@10.1.1.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as3a12d869
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 ACK
    User-Agent: P200 02.09
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Content-Length: 0
    
    
    10 headers, 0 lines
    Destroying call 'CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3'
    
    
    Sip read: 
    ACK sip:**493039833444@10.1.1.1 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T42A7F750
    From: "P200@Asterisk" <sip:p200@10.1.1.1:5060>;tag=00D0E9014CA1_T1218104444
    To: <sip:**493039833444@10.1.1.1:5060>;tag=as3a12d869
    Call-ID: CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3
    CSeq: 684307914 ACK
    User-Agent: P200 02.09
    Contact: <sip:p200@10.1.1.3:5060>
    Max-Forwards: 70
    Content-Length: 0
    
    
    10 headers, 0 lines
    Destroying call 'CALL_ID40_00D0E9014CA1_T325816933@10.1.1.3'
    
     
  4. Maik

    Maik Gesperrt

    Registriert seit:
    1 Apr. 2004
    Beiträge:
    1,778
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Wie ich es mir schon gedacht habe erwartet der snom-Server ein Passwort und da kannst du natuerlich nicht das richtige liefern. Da sollte snom evtl. mal den Server richtig konfigurieren. ;)