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

[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

    Registriert seit:
    31 Jan. 2009
    Beiträge:
    16
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Beruf:
    Programmierer
    Ort:
    Heidesheim am Rhein
    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:[email protected]e.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  [email protected]
           > 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:[email protected]>;tag=as0f2d5b8d
    To: <sip:[email protected]>
    Call-ID: [email protected]
    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:[email protected]: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:[email protected]>;tag=as0f2d5b8d
    To: <sip:[email protected]>;tag=4fa8f7eb71cc68cca91a14abea886308.7c24
    Call-ID: [email protected]
    CSeq: 125 REGISTER
    Contact: <sip:[email protected]:5060>;expires=120
    Content-Length: 0
    
    <------------->
    --- (8 headers 0 lines) ---
    Scheduling destruction of SIP dialog '[email protected]' 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 [[email protected]:1] MSet("DAHDI/i2/73585120-4", "~~EXTEN~~=0151xxxxxxxx") in new stack
        -- Executing [[email protected]:2] Goto("DAHDI/i2/73585120-4", "sw_2_73585120,10") in new stack
        -- Goto (isdntelaus,sw_2_73585120,10)
        -- Executing [[email protected]: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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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 '[email protected]: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:[email protected]192>;tag=as65aec14a
    To: <sip:sipgate.de>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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:[email protected]>;tag=as65aec14a
    To: <sip:sipgate.de>;tag=4fa8f7eb71cc68cca91a14abea886308.7023
    Call-ID: [email protected]:5060
    CSeq: 102 OPTIONS
    Accept: */*
    Accept-Encoding: 
    Accept-Language: en
    Supported: 
    Content-Length: 0
    
    <------------->
    --- (12 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
    Retransmitting #5 (no NAT) to 85.13.156.94:5060:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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 '[email protected]' Method: REGISTER
    Retransmitting #6 (no NAT) to 85.13.156.94:5060:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 78.51.16.192:5060;branch=z9hG4bK6199021c
    Max-Forwards: 70
    From: "73585120" <sip:[email protected]>;tag=as3c3f2588
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]: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 [email protected]: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 '[email protected]:5060' Method: INVITE
    office*CLI> 
    
     
  2. abw1oim

    abw1oim Aktives Mitglied

    Registriert seit:
    26 März 2007
    Beiträge:
    951
    Zustimmungen:
    3
    Punkte für Erfolge:
    18
    Ort:
    Bonn
    Der Dialbefehl passt nicht zur Konfig:
    Code:
    Dial(SIP/sipgate/0151xxxxxxxx,,r)
    wird funktionieren.
     
  3. tameritoke

    tameritoke Neuer User

    Registriert seit:
    31 Jan. 2009
    Beiträge:
    16
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Beruf:
    Programmierer
    Ort:
    Heidesheim am Rhein
    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

    Registriert seit:
    26 März 2007
    Beiträge:
    951
    Zustimmungen:
    3
    Punkte für Erfolge:
    18
    Ort:
    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

    Registriert seit:
    31 Jan. 2009
    Beiträge:
    16
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Beruf:
    Programmierer
    Ort:
    Heidesheim am Rhein
    #5 tameritoke, 9 Sep. 2011
    Zuletzt bearbeitet: 9 Sep. 2011
    Ich habe den Fehler gefunden wie du mir gesagt hast! Fehler im Dialplan!