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In beide Richtungen nichts zu hören

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von D-Trix, 11 Okt. 2005.

  1. D-Trix

    D-Trix Neuer User

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    Ich habe mich heute bei Sipnetworks.de (ehemals Interfoni) angemeldet.

    Registerd ist der Account bereits im asterisk:
    > sip show registry
    > Host Username Refresh State
    > siplogin.de:5060 49180xxxxxxx 1185 Registered

    Bei Anrufversuchen klingelt es zwar, aber nach abheben ist in beiden Richtungen nichts zu hören. Auf den ersten Blick scheint das ja ein Firewall oder NAT- problem zu sein, jedoch funktonieren Sipgate und Sipsnip bereits einwandfrei. Im Sip debug Modus fällt jedoch die Retransmitting Meldung ins Auge:

    Retransmitting #7 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxx8@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK2405cb88;rport
    From: "ABC" <sip:4918051xxxxxx@192.168.0.160>;tag=as1a40237a
    To: <sip:0238xxxxxx@siplogin.de>
    Contact: <sip:49180512xxxxxx@192.168.0.160>
    Call-ID: 6445d95a47d323461d716dfasdc1dfa35482@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Tue, 11 Oct 2005 19:04:44 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265


    aktueller (12.Okt) Auszug der sip.conf:
    Code:
    [general]
    port = 5060
    bindaddr = 192.168.0.160
    ;externip = 212.xxx.xxx.xxx
    localnet = 192.168.0.0/255.255.255.0 :local network and mask
    srvlookup = yes
    context = default
    language=de
    ;Uebertragunsart
    disallow=all
    ;allow=ilbc
    allow=alaw
    allow=ulaw
    allow=g726
    allow=gsm
    allow=slinear
    canreinvite=no ;yes -only works if all devices are on the same side of the NAT
    insecure=very
    nat=yes
    dtmfmode=inband
    ;tos = reliability
    tos=0x18
    maxexpirey = 3600
    defaultexpirey = 1200
    register => SIP-Account-ID:SIP-Passwort@siplogin.de/SIP-Account-ID 
    
    [SIP-Account-ID]
    type=friend
    username=SIP-Account-ID
    fromuser=SIP-Account-ID
    secret=SIP-Passwort
    host=siplogin.de
    fromdomain=siplogin.de
    insecure=very
    nat=no 
    
    Auszug der sip.conf (alt):
    Code:
    [general]
    port = 5060
    bindaddr = 0.0.0.0
    ;externip = 212.xxx.xxx.xxx
    localnet = 192.168.0.0/255.255.255.0 :local network and mask
    srvlookup = yes
    context = default
    language=de
    ;Uebertragunsart
    disallow=all
    ;allow=ilbc
    allow=alaw
    allow=ulaw
    allow=g726
    allow=gsm
    allow=slinear
    canreinvite=no ;yes -only works if all devices are on the same side of the NAT
    insecure=very
    nat=yes
    dtmfmode=inband
    ;tos = reliability
    tos=0x18
    maxexpirey = 3600
    defaultexpirey = 1200
    register => 49180xxxxx:xxxxxx@siplogin.de/sipnetworks
    
    
    
    [sipnetworksout]
    type=friend
    username=49180xxxxxxxxx
    secret=xxxxxxxxx
    host=siplogin.de
    fromuser=49180xxxxxxxxxx
    ;fromdomain=siplogin.de
    context=default
    canreinvite=no
    qualify=no
    ;disallow=all
    disallow=ilbc
    ;allow=alaw
    ;allow=gsm
    insecure=very
    nat=no
    tos=reliability
    

    Was bedeutet das Retransmitting überhaupt und wie kann ich das vermeiden?
     
  2. betateilchen

    betateilchen Grandstream-Guru

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    am Letzenberg
    wieso setzt Du im [general] nat=yes wenn Du es im [sipnetworksout] dann wieder abschaltest ? Bist Du denn nun hinter NAT oder nicht ?
     
  3. D-Trix

    D-Trix Neuer User

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    das stammt noch aus meinen unzähligen probierversuchen. bei sipsnip und sipgate klappt es mit nat=no. ich bin hinter einem isa server 2004 und weiss offen gestanden nicht ob da NAT praktiziert wird oder nicht *schäm* eigentlich müsste es aber doch, da ich mehr pcs im netzwerk habe, als externe IPs?
     
  4. D-Trix

    D-Trix Neuer User

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    Hier noch mal ein SIP-Log von einem "auf-beiden-seiten-lautlos-Gespräch":

    Code:
        -- Executing Dial("CAPI/contr1/99-1", "SIP/0238xxxxxx@sipnetworksout|60") in new stack
    
    
    We're at 192.168.0.160 port 14516
    Answering/Requesting with root capability 0x8 (alaw)
    Answering with preferred capability 0x4 (ulaw)
    Answering with preferred capability 0x10 (g726)
    Answering with preferred capability 0x2 (gsm)
    Answering with preferred capability 0x40 (slin)
    12 headers, 12 lines
    
    Reliably Transmitting (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
        -- Called 0238xxxxxxx@sipnetworksout
    Retransmitting #1 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #2 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #3 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #4 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #5 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #6 (no NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 12:53:59 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15867 15867 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 14516 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Reliably Transmitting (no NAT) to 80.237.199.17:5060:
    CANCEL sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 CANCEL
    User-Agent: Asterisk PBX
    Content-Length: 0
    
    
    ---
    Scheduling destruction of call '50cb821c06ba52454ed853346c33c2fa@192.168.0.160' in 15000 ms
        -- CAPI Hangingup
        -- removed pipe for PLCI = 0x2501
    Retransmitting #1 (no NAT) to 80.237.199.17:5060:
    CANCEL sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxx@192.168.0.160>
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 CANCEL
    User-Agent: Asterisk PBX
    Content-Length: 0
    
    
    ---
    
    <-- SIP read from 80.237.199.17:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.160:5060;received=212.132.194.xx;received=212.132.194.xx;branch=z9hG4bK7d72a53b;rport=58350
    From: "AB" <sip:49180xxxxxxx@192.168.0.160>;tag=as5405c760
    To: <sip:0238xxxxxxx@siplogin.de>;tag=as461cef1f
    Call-ID: 50cb821c06ba52454ed853346c33c2fa@192.168.0.160
    CSeq: 102 CANCEL
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Contact: <sip:0238xxxxxxx@80.237.199.3>
    Content-Length: 0
    
    
     
  5. D-Trix

    D-Trix Neuer User

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    Hier das ganze nochmal mit nat=yes :

    Code:
    -- Executing Dial("CAPI/contr1/99-0", "SIP/0238xxxxxxx@sipnetworksout|60") in new stack
    We're at 192.168.0.160 port 15834
    Answering/Requesting with root capability 0x8 (alaw)
    Answering with preferred capability 0x4 (ulaw)
    Answering with preferred capability 0x10 (g726)
    Answering with preferred capability 0x2 (gsm)
    Answering with preferred capability 0x40 (slin)
    12 headers, 12 lines
    Reliably Transmitting (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
        -- Called 0238xxxxxxx@sipnetworksout
    Retransmitting #1 (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #2 (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #3 (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #4 (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Retransmitting #5 (NAT) to 80.237.199.17:5060:
    INVITE sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Wed, 12 Oct 2005 13:04:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 265
    
    v=0
    o=root 15916 15916 IN IP4 192.168.0.160
    s=session
    c=IN IP4 192.168.0.160
    t=0 0
    m=audio 15834 RTP/AVP 8 0 111 3 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    ---
    Reliably Transmitting (NAT) to 80.237.199.17:5060:
    CANCEL sip:0238xxxxxxx@siplogin.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>
    Contact: <sip:49180xxxxxxxxxx@192.168.0.160>
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 CANCEL
    User-Agent: Asterisk PBX
    Content-Length: 0
    
    
    ---
    Scheduling destruction of call '1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160' in 15000 ms
        -- CAPI Hangingup
        -- removed pipe for PLCI = 0x2601
    
    <-- SIP read from 80.237.199.17:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.160:5060;received=212.132.194.xx;received=212.132.194.xx;branch=z9hG4bK4dd9229c;rport=58266
    From: "AB" <sip:49180xxxxxxxxxx@192.168.0.160>;tag=as5ab37a23
    To: <sip:0238xxxxxxx@siplogin.de>;tag=as1a790d3e
    Call-ID: 1b6dc95a68153fra26ecb42a5601cf80@192.168.0.160
    CSeq: 102 CANCEL
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Contact: <sip:0238xxxxxxx@80.237.199.3>
    Content-Length: 0
    
    
    --- (10 headers 0 lines)---
    

    Werden noch weitere Informationen gebraucht?


    /edit: ich habs jetzt mal in der anderen Richtung versucht, ruft man über die sipnetworks nummer (1805xxxxxxxxxxx) Asterisk an, kann man den anrufer hören, umgekehrt jedoch nix
     
  6. VoIP-noob

    VoIP-noob Neuer User

    Registriert seit:
    16 März 2005
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    Nunja, eigentlich gibts den Laden schon länger als Interfoni ... wollte ich nur mal bemerken.

    Ich kenn mich zwar nicht so mit Asterisk aus, aber die einseitige Sprachverbindung hört sich für mich irgendwie nach einem NAT-Problem an.