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Festnetz über Freenet: * ignoriert RTP von Freenet

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von saddamski, 19 Sep. 2006.

  1. saddamski

    saddamski Neuer User

    Registriert seit:
    28 Apr. 2006
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    Hallo zusammen!
    Ich habe leider ein Problem mit * im Zusammenhang mit dem Freenet ISDN Gateway.

    Der Aufbau ist folgender:
    DECT Telefon -> isdn -> ITA -> ethernet -> Asterisk -> ethernet -> Router ->Internet -> Freenet -> isdn -> Mein Handy.

    Dabei funktioniert die Audioverbindung von Handy zu DECT nicht. Die andere Richtung funktioniert tadellos. Das interessante dabei ist, dass die RTP Daten von Freenet aber den Asterisk erreichen. Nur scheint dieser die zu verwerfen bzw. zu ignorieren.
    Ansonsten funktioniert alles. Auch Gespräche über das Internet funktionieren. Mit Sipgate habe ich z.B. keine Probleme. Was mir dabei aufgefallen ist, dass bei Sipgate die IP von SIP Proxy und ISDN Gateway gleich sind. Bei Freenet sind diese verschieden. Ob das ein Grund ist?
    Ich benutze die Version 1.2.7.1 auf einer Debian Kiste.
    Habe auch mal im Bugtracker gesucht, allerdings nix passendes gefunden. Wobei ich mir da auch sicher bin dass ich nach evtl. nach den falschen Wörtern gesucht habe.


    Hier mal die ganzen Einstellungen und Logs

    tcpdump
    Code:
    13:41:59.660438 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.668426 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.668621 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.680816 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.688520 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.688691 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.695860 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.708596 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.708789 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.715952 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.728497 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.728682 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.736311 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.748371 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.748566 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.756083 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.768400 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.768597 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.776580 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    13:41:59.788487 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
    13:41:59.788657 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
    13:41:59.796765 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
    
    rtp debug ip 62.104.212.131 in der Asterisl CLI zeigt:
    Code:
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46095, ts 372000, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46096, ts 372160, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46097, ts 372320, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46098, ts 372480, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46099, ts 372640, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46100, ts 372800, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46101, ts 372960, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46102, ts 373120, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46103, ts 373280, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46104, ts 373440, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46105, ts 373600, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46106, ts 373760, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46107, ts 373920, len 160)
    Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46108, ts 374080, len 160)
    
    sip show channels während das Gespräch aufgebaut ist:
    Code:
    Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
    194.97.54.97     0173xxxxxx  5d84bf0d37b  00103/00000  alaw  No       Tx: ACK
    192.168.0.6      132         0017364b04-  00101/05889  alaw  No       Rx: ACK
    
    Die Ausgabe in der CLI:
    Code:
        -- Executing Dial("SIP/132-b3af", "SIP/0173xxxxxxx@Freenet_THO|45|") in new stack
        -- Called 0173xxxxxxx@Freenet_THO
        -- SIP/Freenet_THO-cc1e is ringing
        -- SIP/Freenet_THO-cc1e is making progress passing it to SIP/132-b3af
        -- SIP/Freenet_THO-cc1e answered SIP/132-b3af
        -- Attempting native bridge of SIP/132-b3af and SIP/Freenet_THO-cc1e
      == Spawn extension (Tel132, 0173xxxxxxx, 1) exited non-zero on 'SIP/132-b3af'
    
    sip.conf
    Code:
    [general]
    context=default           
    bindport=5060             
    bindaddr=0.0.0.0          
    srvlookup=yes             
    defaultexpirey=1800
    language=de
    disallow=all
    allow=alaw
    allow=ulaw
    externhost=xxx.dyndns.org
    localnet=192.168.0.0/255.255.255.0
    nat=yes
    ;---------------------------------------------------------
    ;Sipgate xxxx
    register => 65XXXXX:xxxsipgate.de/Sipgate_TVA
    
    ;Sipgate xxxx
    register => 19xxxx:xxx@sipgate.de/Sipgate_THO
    
    ;Freenet xxx
    register => Elxxxxxx:xxxxxxx@freenet.de/Freenet_THO
    
    [Sipgate_TVA]
    type=peer
    username=65xxxx
    fromuser=65xxxx
    secret=xxx
    host=sipgate.de
    fromdomain=sipgate.de
    insecure=very
    canreinvite=no
    nat=yes
    
    [Sipgate_THO]
    type=peer
    username=19xxx
    fromuser=19xxx
    secret=xxxx
    host=sipgate.de
    fromdomain=sipgate.de
    insecure=very
    canreinvite=no
    nat=yes
    
    [Freenet_THO]
    type=friend
    username=Elxxxx
    fromuser=Elxxx
    secret=xxx
    host=freenet.de
    fromdomain=freenet.de
    insecure=very
    canreinvite=no
    nat=yes
    port=5060
    
    [freenet_in]
    type=friend
    fromdomain=freenet.de
    host=freenet.de
    context=ankommend
    nat=yes
    ;qualify=yes
    canreinvite=no
    
    Hier folgen dann noch diverse Endgeräte
    
    Ausschnitt aus der extensions.conf
    Code:
    [freenet_out_THO]
    exten => _0XXX.,1,Dial,SIP/${EXTEN}@Freenet_THO|45|
    
    sip debug peer Freenet_THO
    Code:
    We're at 80.130.237.89 port 12048
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    13 headers, 11 lines
    Reliably Transmitting (NAT) to 194.97.54.97:5060:
    INVITE sip:0173xxxxxxx@freenet.de SIP/2.0
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>
    Contact: <sip:Elxxxx@80.130.237.89>
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    Seq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Tue, 19 Sep 2006 11:58:14 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 240
    
    v=0
    o=root 3018 3018 IN IP4 80.130.237.89
    s=session
    c=IN IP4 80.130.237.89
    t=0 0
    m=audio 12048 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=silenceSupp:off - - - -
    
    ---
        -- Called 0173xxxxxxx@Freenet_THO
    asterisk*CLI>
    <-- SIP read from 194.97.54.97:5060:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport=5060
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>;tag=36a431905c116a7e58f13c316b42f021.3910
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest realm="freenet.de", nonce="450fdc81e3797e48848abf485651594b42f30064", qop="auth"
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    Transmitting (NAT) to 194.97.54.97:5060:
    ACK sip:0173xxxxxxx@freenet.de SIP/2.0
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>;tag=36a431905c116a7e58f13c316b42f021.3910
    Contact: <sip:Elxxxx@80.130.237.89>
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    We're at 80.130.237.89 port 12048
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 194.97.54.97:5060:
    INVITE sip:0173xxxxxxx@freenet.de SIP/2.0
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>
    Contact: <sip:Elxxxx@80.130.237.89>
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Proxy-Authorization: Digest username="Elxxxx", realm="freenet.de", algorithm=MD5, uri="sip:0173xxxxxxx@freenet.de", nonce="xxxxxx", response="xxxxxxx", opaque="", qop=auth, cnonce="1b7c5116", nc=00000001
    Date: Tue, 19 Sep 2006 11:58:14 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 240
    
    v=0
    o=root 3018 3019 IN IP4 80.130.237.89
    s=session
    c=IN IP4 80.130.237.89
    t=0 0
    m=audio 12048 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=silenceSupp:off - - - -
    
    ---
    asterisk*CLI>
    <-- SIP read from 194.97.54.97:5060:
    SIP/2.0 100 trying -- your call is important to us
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 103 INVITE
    Content-Length: 0
    
    
    --- (7 headers 0 lines)---
    asterisk*CLI>
    <-- SIP read from 194.97.54.97:5060:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>;tag=008082384A4502248FAE00000F77
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 103 INVITE
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
    Content-Type: application/sdp
    Content-Length:   183
    
    v=0
    o=- 3106003 0 IN IP4 62.104.212.131
    s=session
    c=IN IP4 62.104.212.131
    t=0 0
    m=audio 19192 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    
    --- (9 headers 9 lines)---
    Found RTP audio format 8
    Found RTP audio format 101
    Peer audio RTP is at port 62.104.212.131:19192
    Found description format PCMA
    Found description format telephone-event
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
        -- SIP/Freenet_THO-af80 is ringing
        -- SIP/Freenet_THO-af80 is making progress passing it to SIP/132-b6e5
    asterisk*CLI>
    <-- SIP read from 194.97.54.97:5060:
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
    Record-Route: <sip:194.97.59.151;ftag=as1566d63f;lr=on>
    Record-Route: <sip:194.97.54.97;ftag=as1566d63f;lr=on>
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>;tag=008082384A4502248FAE00000F77
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 103 INVITE
    Contact: <sip:195.4.21.244:5060>
    Allow-Events: refer
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
    Content-Type: application/sdp
    Supported: 100rel, timer, replaces
    Content-Length:   183
    
    v=0
    o=- 3106003 0 IN IP4 62.104.212.131
    s=session
    c=IN IP4 62.104.212.131
    t=0 0
    m=audio 19192 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    
    --- (14 headers 9 lines)---
    Found RTP audio format 8
    Found RTP audio format 101
    Peer audio RTP is at port 62.104.212.131:19192
    Found description format PCMA
    Found description format telephone-event
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    list_route: hop: <sip:194.97.54.97;ftag=as1566d63f;lr=on>
    list_route: hop: <sip:194.97.59.151;ftag=as1566d63f;lr=on>
    set_destination: Parsing <sip:194.97.54.97;ftag=as1566d63f;lr=on> for address/port to send to
    set_destination: set destination to 194.97.54.97, port 5060
    Transmitting (NAT) to 194.97.54.97:5060:
    ACK sip:195.4.21.244:5060 SIP/2.0
    Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK02af255f;rport
    Route: <sip:194.97.54.97;ftag=as1566d63f;lr=on>,<sip:194.97.59.151;ftag=as1566d63f;lr=on>
    From: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    To: <sip:0173xxxxxxx@freenet.de>;tag=008082384A4502248FAE00000F77
    Contact: <sip:Elxxxx@80.130.237.89>
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 103 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
        -- SIP/Freenet_THO-af80 answered SIP/132-b6e5
        -- Attempting native bridge of SIP/132-b6e5 and SIP/Freenet_THO-af80
    asterisk*CLI>
    <-- SIP read from 194.97.54.97:5060:
    BYE sip:Elxxxx@80.130.237.89 SIP/2.0
    Record-Route: <sip:194.97.54.97;ftag=008082384A4502248FAE00000F77;lr=on>
    Record-Route: <sip:194.97.59.151;ftag=008082384A4502248FAE00000F77;lr=on>
    Via: SIP/2.0/UDP 194.97.54.97;branch=0
    Via: SIP/2.0/UDP 194.97.59.151;branch=z9hG4bK5e65.cc686e41.0
    Via: SIP/2.0/UDP 195.4.21.244:5060;branch=z9hG4bK008082384A4502249639000012F8
    From: <sip:0173xxxxxxx@freenet.de>;tag=008082384A4502248FAE00000F77
    To: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 8543 BYE
    Contact: <sip:0173xxxxxxx@195.4.21.244:5060>
    Max-Forwards: 15
    Content-Length:     0
    
    
    --- (13 headers 0 lines)---
    Sending to 194.97.54.97 : 5060 (NAT)
    Transmitting (NAT) to 194.97.54.97:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 194.97.54.97;branch=0;received=194.97.54.97
    Via: SIP/2.0/UDP 194.97.59.151;branch=z9hG4bK5e65.cc686e41.0
    Via: SIP/2.0/UDP 195.4.21.244:5060;branch=z9hG4bK008082384A4502249639000012F8
    ecord-Route: <sip:194.97.54.97;ftag=008082384A4502248FAE00000F77;lr=on>
    Record-Route: <sip:194.97.59.151;ftag=008082384A4502248FAE00000F77;lr=on>
    From: <sip:0173xxxxxxx@freenet.de>;tag=008082384A4502248FAE00000F77
    To: "131" <sip:Elxxxx@freenet.de>;tag=as1566d63f
    Call-ID: 3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de
    CSeq: 8543 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:Elxxxx@80.130.237.89>
    Content-Length: 0
    X-Asterisk-HangupCause: Normal Clearing
    
    
    ---
      == Spawn extension (Tel132, 0173xxxxxxx, 1) exited non-zero on 'SIP/132-b6e5'
    Destroying call '3a75e39e2d2c28860dbbae2d5cdd70ce@freenet.de'
    asterisk*CLI> sip no debug
    
    
    Gibts jemanden der dazu eine Idee hat?
    Habe beim Asterisk auch mal mit dem ethereal geloggt.
    Die RTP's von Freenet kommen wirklich an und sind auch korrekt. Wenn ich die RTPs im ethereal extrahiere kann ich mir die auch anhören. Die Audiodaten kommen also ohne Probleme. Ansonsten funktioniert der Betrieb ja auch..nur Freenet macht ärger.
    Ach ja..in der /etc/hosts ist der Eintrag "194.97.54.97 freenet.de" drin.

    Tobias
     
  2. saddamski

    saddamski Neuer User

    Registriert seit:
    28 Apr. 2006
    Beiträge:
    5
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Niemand der eine Idee hat?
    Sind die Einstellungen denn wenigstens richtig? Es würde mir schon sehr helfen wenn ich wüssste dass es nicht an mir liegt :D
    Die Einstellungen habe ich aus dem Kurs angepasst. Hier und da sind ein paar Sachen aus dem Forum abgeguckt. Also eigentlich sollte die Konfiguration laufen...zumindest tut sie es anscheinend bei anderen.

    Gruß, Tobias