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FreePBX und Asterisk keine Abgehenden Gespräch

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von Monotron, 26 Okt. 2006.

  1. Monotron

    Monotron Neuer User

    Registriert seit:
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    #1 Monotron, 26 Okt. 2006
    Zuletzt bearbeitet: 26 Okt. 2006
    Also ich bin hier schon ewig am herumdoktorn. Aber ich kann nicht raus telefonieren.

    Das Telefon gibt immer ein "Fehler 404" wieder.
    Ich hab keine Ahnung. Würde mich freuen wenn mir da einer helfen könnte.
    Wie gesagt, es geht um FreePBX und Asterisk.

    Intern und ankommend ist alles sauber!
    Habe einmal Arcor für abgehend und ankommmend und nochmal Sipgate für ankommende Calls.

    Hier wie ich die Arcor Trunk eingerichtet habe:

    Code:
    [B]General Settings[/B]
    Outbound Caller ID: 
    Maximum Channels:
    
    [B]Outgoing Dial Rules[/B]
    Dial Rules:
    Outbound Dial Prefix:
    
    [B]Outgoing Settings[/B]
    Trunk Name: arcor
    Peer Details: canreinvite=no
                      fromuser=02115*****
                      host=0211.sip.arcor.de
                      nat=no
                      secret=********
                      username=02115******
    
    [B]Incoming Settings[/B]
    User Context: 02115******
    User Details: allow=ulaw
                      context=ankommend
                      diallow=all
                      fromdomain=0211.sip.arcor.de
                      host=0211.sip.arcor.de
    
    [B]Registration[/B]
    02115*****:******@0211.sip.arcor.de
    
    Und hier dann wie ich die Outbound Route eingerichtet habe:

    Code:
    RouteName: ArcorOut
    Route Password:
    Emergency Callking:
    Dial Patterns: 0.
    Trunk Sequence: 0 SIP/arcor
    
    Hier mal das Ergebnis einen Abgehenden Calls im Sip Debug Modus:

    Code:
    Peer audio RTP is at port 192.168.1.2:49406
    Found description format GSM
    Found description format iLBC
    Found description format iLBC
    Found description format PCMA
    Found description format PCMU
    Found description format telephone-event
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Looking for 02119****** in from-internal (domain 192.168.1.5)
    list_route: hop: <sip:100@192.168.1.2:5060>
    Transmitting (no NAT) to 192.168.1.2:5060:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
    From: "unknown"<sip:100@192.168.1.5>;tag=9498910925858
    To: <sip:02119******@192.168.1.5>
    Call-ID: B5832725-3FE7-4A69-BDC1-D1981E3D943D@192.168.1.2
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:02119******@192.168.1.5>
    Content-Length: 0
    
    
    ---
        -- Executing Macro("SIP/100-081d0570", "dialout-trunk|2|02119******||") in new stack
        -- Executing Set("SIP/100-081d0570", "DIAL_TRUNK=2") in new stack
        -- Executing Set("SIP/100-081d0570", "DIAL_NUMBER=02119******") in new stack
        -- Executing Set("SIP/100-081d0570", "ROUTE_PASSWD=") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "1?noauth") in new stack
        -- Goto (macro-dialout-trunk,s,6)
        -- Executing Set("SIP/100-081d0570", "GROUP()=OUT_2") in new stack
        -- Executing Macro("SIP/100-081d0570", "user-callerid") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "0?report") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "0?start") in new stack
        -- Executing Set("SIP/100-081d0570", "REALCALLERIDNUM=100") in new stack
        -- Executing NoOp("SIP/100-081d0570", "REALCALLERIDNUM is 100") in new stack
        -- Executing Set("SIP/100-081d0570", "AMPUSER=100") in new stack
        -- Executing Set("SIP/100-081d0570", "AMPUSERCIDNAME=Cyraxx") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "0?report") in new stack
        -- Executing Set("SIP/100-081d0570", "CALLERID(all)=Cyraxx <100>") in new stack
        -- Executing NoOp("SIP/100-081d0570", "Using CallerID "Cyraxx" <100>") in new stack
        -- Executing Macro("SIP/100-081d0570", "record-enable|100|OUT") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "0 > 0?2:4") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing AGI("SIP/100-081d0570", "recordingcheck|20061026-085337|1161845617.395") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20061026-085337|1161845617.395: Outbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing NoOp("SIP/100-081d0570", "No recording needed") in new stack
        -- Executing Macro("SIP/100-081d0570", "outbound-callerid|2") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "1?start") in new stack
        -- Goto (macro-outbound-callerid,s,3)
        -- Executing NoOp("SIP/100-081d0570", "REALCALLERIDNUM is 100") in new stack
        -- Executing Set("SIP/100-081d0570", "USEROUTCID=") in new stack
        -- Executing Set("SIP/100-081d0570", "EMERGENCYCID=") in new stack
        -- Executing Set("SIP/100-081d0570", "TRUNKOUTCID=") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,11)
        -- Executing GotoIf("SIP/100-081d0570", "1?usercid") in new stack
        -- Goto (macro-outbound-callerid,s,13)
        -- Executing GotoIf("SIP/100-081d0570", "1?report") in new stack
        -- Goto (macro-outbound-callerid,s,17)
        -- Executing NoOp("SIP/100-081d0570", "CallerID set to "Cyraxx" <100>") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "1?nomax") in new stack
        -- Goto (macro-dialout-trunk,s,12)
        -- Executing AGI("SIP/100-081d0570", "fixlocalprefix") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
      fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
        -- AGI Script fixlocalprefix completed, returning 0
        -- Executing Set("SIP/100-081d0570", "OUTNUM=02119843883") in new stack
        -- Executing Set("SIP/100-081d0570", "custom=SIP/arcor") in new stack
        -- Executing GotoIf("SIP/100-081d0570", "0?customtrunk") in new stack
        -- Executing Dial("SIP/100-081d0570", "SIP/arcor/02119*******|120|r") in new stack
    Destroying call '763e30406c058eac718fe4931a8e30a9@192.168.1.5'
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Executing Goto("SIP/100-081d0570", "s-CHANUNAVAIL|1") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
        -- Executing NoOp("SIP/100-081d0570", "Dial failed due to CHANUNAVAIL") in new stack
        -- Executing Macro("SIP/100-081d0570", "outisbusy|") in new stack
        -- Executing Playback("SIP/100-081d0570", "all-circuits-busy-now|noanswer") in new stack
    We're at 192.168.1.5 port 13182
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Transmitting (no NAT) to 192.168.1.2:5060:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
    From: "unknown"<sip:100@192.168.1.5>;tag=9498910925858
    To: <sip:02119*******@192.168.1.5>;tag=as2183240d
    Call-ID: B5832725-3FE7-4A69-BDC1-D1981E3D943D@192.168.1.2
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:02119*******@192.168.1.5>
    Content-Type: application/sdp
    Content-Length: 236
    
    v=0
    o=root 5941 5941 IN IP4 192.168.1.5
    s=session
    c=IN IP4 192.168.1.5
    t=0 0
    m=audio 13182 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=silenceSupp:off - - - -
    
    ---
        -- Playing 'all-circuits-busy-now' (language 'en')
    REGISTER 13 headers, 0 lines
    Reliably Transmitting (no NAT) to 212.144.24.22:5060:
    REGISTER sip:0211.sip.arcor.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK02f8b5fd;rport
    From: <sip:02115******@0211.sip.arcor.de>;tag=as6dbbcb90
    To: <sip:02115******@0211.sip.arcor.de>
    Call-ID: 6300b3df48d228317aff1a21637065b3@192.168.1.5
    CSeq: 107 REGISTER
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Authorization: Digest username="021156*****", realm="arcor.de", algorithm=MD5, uri="sip:0211.sip.arcor.de", nonce="45407704f08854d6ad33f538b55183b9a0e8b54c", response="bdf4b1d34b3d00ea496166f2abbe186a", opaque=""
    Expires: 120
    Contact: <sip:s@192.168.1.5>
    Event: registration
    Content-Length: 0
    
    
    ---
    asterisk*CLI>
    <-- SIP read from 212.144.24.22:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.5:5060;received=88.76.211.55;branch=z9hG4bK02f8b5fd;rport=5060
    From: <sip:02115*******@0211.sip.arcor.de>;tag=as6dbbcb90
    To: <sip:02115*******@0211.sip.arcor.de>
    Call-ID: 6300b3df48d228317aff1a21637065b3@192.168.1.5
    CSeq: 107 REGISTER
    Contact: <sip:s@192.168.1.5>;expires=60
    
    
    --- (7 headers 0 lines) ---
    Scheduling destruction of call '6300b3df48d228317aff1a21637065b3@192.168.1.5' in 32000 ms
        -- Executing Playback("SIP/100-081d0570", "pls-try-call-later|noanswer") in new stack
        -- Playing 'pls-try-call-later' (language 'en')
    asterisk*CLI>
    <-- SIP read from 192.168.1.2:5060:
    OPTIONS sip:192.168.1.5:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a8010200000010454077a5000026e200002c36
    Content-Length: 0
    Call-ID: 2CBA17BC-E844-41A6-A0E6-D2EF7DFA674C@192.168.1.2
    CSeq: 3303 OPTIONS
    From: <sip:100@192.168.1.5>;tag=9499321827002
    Max-Forwards: 70
    To: <sip:192.168.1.5:5060>
    
    
    --- (8 headers 0 lines) ---
    Looking for s in from-sip-external (domain 192.168.1.5)
    Transmitting (no NAT) to 192.168.1.2:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a8010200000010454077a5000026e200002c36;received=192.168.1.2
    From: <sip:100@192.168.1.5>;tag=9499321827002
    To: <sip:192.168.1.5:5060>;tag=as22058826
    Call-ID: 2CBA17BC-E844-41A6-A0E6-D2EF7DFA674C@192.168.1.2
    CSeq: 3303 OPTIONS
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:192.168.1.5>
    Accept: application/sdp
    Content-Length: 0
    
    
    ---
    Destroying call '2CBA17BC-E844-41A6-A0E6-D2EF7DFA674C@192.168.1.2'
        -- Executing Macro("SIP/100-081d0570", "hangupcall") in new stack
        -- Executing ResetCDR("SIP/100-081d0570", "w") in new stack
        -- Executing NoCDR("SIP/100-081d0570", "") in new stack
        -- Executing Wait("SIP/100-081d0570", "5") in new stack
    asterisk*CLI>
    <-- SIP read from 192.168.1.3:5060:
    
    
    --- (0 headers 0 lines) Nat keepalive ---
        -- Executing Hangup("SIP/100-081d0570", "") in new stack
      == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570' in macro 'hangupcall'
      == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570' in macro 'outisbusy'
      == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570'
    Scheduling destruction of call 'B5832725-3FE7-4A69-BDC1-D1981E3D943D@192.168.1.2' in 32000 ms
    Reliably Transmitting (no NAT) to 192.168.1.2:5060:
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
    From: "unknown"<sip:100@192.168.1.5>;tag=9498910925858
    To: <sip:02119*******@192.168.1.5>;tag=as2183240d
    Call-ID: B5832725-3FE7-4A69-BDC1-D1981E3D943D@192.168.1.2
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0
    
    
    ---
    asterisk*CLI>
    <-- SIP read from 192.168.1.2:5060:
    ACK sip:02119*******@192.168.1.5 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33
    Content-Length: 0
    Call-ID: B5832725-3FE7-4A69-BDC1-D1981E3D943D@192.168.1.2
    CSeq: 2 ACK
    From: "unknown"<sip:100@192.168.1.5>;tag=9498910925858
    Max-Forwards: 70
    To: <sip:02119843883@192.168.1.5>;tag=as2183240d
    User-Agent: SJphone/1.60.289a (SJ Labs)
    
    
    Wenn ihr mehr Infos bräuchtet, sagt mir bitte was, ich verzweifle hier noch :-(
     
  2. Monotron

    Monotron Neuer User

    Registriert seit:
    12 Okt. 2006
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    Schade das nichts als Antwort kam.
    Aber ich habe die Lösung mitlerweile auch gefunden.

    In der Trunk bei Outbound folgendes einstellen, dann gehts:

    Code:
    allow=gsm&ilbc&alaw&ulaw
    context=from-pstn
    disallow=all
    fromdomain=0211.sip.arcor.de
    fromuser=02115*******
    host=0211.sip.arcor.de
    insecure=very
    secret=********
    type=friend
    username=02115*******