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[Problem] kann keine Verbindung asterisk 1.8.6 und sipgate herstellen.

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von tameritoke, 8 Sep. 2011.

  1. tameritoke

    tameritoke Neuer User

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    Hi!
    Ich schaffe es nicht, mit sipgate erfolgreich eine Verbindung zu einem anderem Teilnehmer aufzubauen.

    Die Linux Maschine baut die Verbindung selbst per PPPoE auf, also es befindet sich auch keine Firewall oder sonstiger NAT Router dazwischen.

    Ich komme jedenfalls nicht dahinter, wie ich dieses Problem gelöst kriege. Unter Asterisk 1.6.0 lief es, nur stehe ich jetzt total auf dem Schlauch.

    Für jede Hilfe bin ich dankbar.

    sip.conf:
    Code:
    [general]
    port=5060
    bindaddr=0.0.0.0
    language=de
    dtmfmode=rfc2833
    srvlookup=yes
    disallow=all
    allow=alaw
    allow=ulaw
    nat=no
    
    register = 334xxxxxxx:xxxxxxx@sipgate.de/2000
    
    [sipgate]
    type=friend
    insecure=invite
    username=334xxxxxxx
    fromuser=334xxxxxxx
    fromdomain=sipgate.de
    host=sipgate.de
    secret=xxxxxxx
    host=sipgate.de
    qualify=yes
    caninvite=no
    canreinvite=no
    disallow=all
    allow=alaw
    allow=ulaw
    allow=speex
    allow=g729
    dtmfmode=rfc2833
    context=tamertelein
    

    und der sip.debug:

    Code:
    office*CLI> sip set debug on
    SIP Debugging re-enabled
        -- Accepting overlap call from '73585120' to '0151xxxxxxxx' on channel 0/2, span 2
        -- Starting simple switch on 'DAHDI/i2/73585120-4'
    [Sep  8 23:28:26] NOTICE[17120]: chan_sip.c:12593 sip_reregister:    -- Re-registration for  334xxxxxxx@sipgate.de
           > doing dnsmgr_lookup for 'sipgate.de'
           > ast_get_srv: SRV lookup for '_sip._udp.sipgate.de' mapped to host sipgate.de, port 5060
    REGISTER 11 headers, 0 lines
    Reliably Transmitting (no NAT) to 217.10.79.9:5060:
    REGISTER sip:sipgate.de SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK2fec2dd6
    Max-Forwards: 70
    From: <sip:334xxxxxxx@sipgate.de>;tag=as0f2d5b8d
    To: <sip:334xxxxxxx@sipgate.de>
    Call-ID: 3c7ffdf762852f284c36458c0f0d6b69@127.0.0.1
    CSeq: 125 REGISTER
    User-Agent: Asterisk PBX 1.8.6.0
    Authorization: Digest username="334xxxxxxx", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="4e6933d47deb3e2eb7989ee2e3c20a2f8462dc94", response="672e5f7f64c9936f974cd679bb6552e2"
    Expires: 120
    Contact: <sip:06132xxxxxxx@78.51.16.192:5060>
    Content-Length: 0
    
    
    ---
    
    <--- SIP read from UDP:217.10.79.9:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK2fec2dd6
    From: <sip:334xxxxxxx@sipgate.de>;tag=as0f2d5b8d
    To: <sip:334xxxxxxx@sipgate.de>;tag=4fa8f7eb71cc68cca91a14abea886308.7c24
    Call-ID: 3c7ffdf762852f284c36458c0f0d6b69@127.0.0.1
    CSeq: 125 REGISTER
    Contact: <sip:06132xxxxxxx@78.51.16.192:5060>;expires=120
    Content-Length: 0
    
    <------------->
    --- (8 headers 0 lines) ---
    Scheduling destruction of SIP dialog '3c7ffdf762852f284c36458c0f0d6b69@127.0.0.1' in 32000 ms (Method: REGISTER)
    [Sep  8 23:28:26] NOTICE[17120]: chan_sip.c:20125 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
        -- Executing [0151xxxxxxxx@isdntelaus:1] MSet("DAHDI/i2/73585120-4", "~~EXTEN~~=0151xxxxxxxx") in new stack
        -- Executing [0151xxxxxxxx@isdntelaus:2] Goto("DAHDI/i2/73585120-4", "sw_2_73585120,10") in new stack
        -- Goto (isdntelaus,sw_2_73585120,10)
        -- Executing [sw_2_73585120@isdntelaus:10] Dial("DAHDI/i2/73585120-4", "SIP/334xxxxxxx/0151xxxxxxxx,,r") in new stack
      == Using SIP RTP CoS mark 5
    Audio is at 5060
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
        -- Called SIP/334xxxxxxx/0151xxxxxxxx
    Retransmitting #1 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
    Retransmitting #2 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
    Retransmitting #3 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
    Retransmitting #4 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
        -- Span 2: Channel 0/2 got hangup request, cause 16
    Scheduling destruction of SIP dialog '56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060' in 32000 ms (Method: INVITE)
      == Spawn extension (isdntelaus, sw_2_73585120, 10) exited non-zero on 'DAHDI/i2/73585120-4'
        -- Hungup 'DAHDI/i2/73585120-4'
    Reliably Transmitting (no NAT) to 217.10.79.9:5060:
    OPTIONS sip:sipgate.de SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK76414742
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@78.51.16.192>;tag=as65aec14a
    To: <sip:sipgate.de>
    Contact: <sip:asterisk@78.51.16.192:5060>
    Call-ID: 05ed387d6b261e07592c809a2f6832ed@78.51.16.192:5060
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:39 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0
    
    
    ---
    
    <--- SIP read from UDP:217.10.79.9:5060 --->
    SIP/2.0 200 OK
    Record-Route: <sip:217.10.79.9;lr=on;ftag=as65aec14a>
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK76414742
    From: "asterisk" <sip:asterisk@78.51.16.192>;tag=as65aec14a
    To: <sip:sipgate.de>;tag=4fa8f7eb71cc68cca91a14abea886308.7023
    Call-ID: 05ed387d6b261e07592c809a2f6832ed@78.51.16.192:5060
    CSeq: 102 OPTIONS
    Accept: */*
    Accept-Encoding: 
    Accept-Language: en
    Supported: 
    Content-Length: 0
    
    <------------->
    --- (12 headers 0 lines) ---
    Really destroying SIP dialog '05ed387d6b261e07592c809a2f6832ed@78.51.16.192:5060' Method: OPTIONS
    Retransmitting #5 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
    Really destroying SIP dialog '3c7ffdf762852f284c36458c0f0d6b69@127.0.0.1' Method: REGISTER
    Retransmitting #6 (no NAT) to 85.13.156.94:5060:
    INVITE sip:0151xxxxxxxx@334xxxxxxx SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:73585120@78.51.16.192>;tag=as3c3f2588
    To: <sip:0151xxxxxxxx@334xxxxxxx>
    Contact: <sip:73585120@78.51.16.192:5060>
    Call-ID: 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Thu, 08 Sep 2011 21:28:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260
    
    v=0
    o=root 1297421477 1297421477 IN IP4 78.51.16.192
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 78.51.16.192
    t=0 0
    m=audio 28072 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
    [Sep  8 23:28:59] WARNING[17120]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission 56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32000ms with no response
    Really destroying SIP dialog '56e79b637ad5395d2b994003178d35bd@78.51.16.192:5060' Method: INVITE
    office*CLI> 
    
     
  2. abw1oim

    abw1oim Aktives Mitglied

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    Bonn
    Der Dialbefehl passt nicht zur Konfig:
    Code:
    Dial(SIP/sipgate/0151xxxxxxxx,,r)
    wird funktionieren.
     
  3. tameritoke

    tameritoke Neuer User

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    Der DIAL Befehl stimmt, du irsst dich. Ausserdem, wenn der nicht stimmen würde, käme die Meldung "extension not found" oder was ähnliches und ich hätte keinen sip debug. Du siehst ja selber im debug, dass asterisk die Pakete nicht weiterleiten kann.

    Schaue dir den debug genau an! Morgen lese ich mir mal die SIP RFC komplett durch. Irgendwo muss ja der Fehler liegen.
     
  4. abw1oim

    abw1oim Aktives Mitglied

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    Bonn
    Sorry, aber das ist Unsinn! Das ist ein abgehender Call und ein Lesen des SIP-Debugs würde Dir aufzeigen, dass versucht wird, eine Verbindung zur IP 334xxxxxxx aufzubauen, die natürlich nicht existiert (da sie über die Konfig nicht aufgelöst werden kann) und daher permanant retransmitted wird.
    Und "extension not found" ist nur bei eingehenden Calls relevant!
     
  5. tameritoke

    tameritoke Neuer User

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    #5 tameritoke, 9 Sep. 2011
    Zuletzt bearbeitet: 9 Sep. 2011
    Ich habe den Fehler gefunden wie du mir gesagt hast! Fehler im Dialplan!