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Sipgate raustelefonieren geht, angerufen werden kann ich nic

Dieses Thema im Forum "Asterisk Rufnummernplan" wurde erstellt von BlackSektor, 3 Dez. 2004.

  1. BlackSektor

    BlackSektor Neuer User

    Registriert seit:
    3 Nov. 2004
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    Hallo, ich kann nun mit meinem Asterisk raustelefonieren (über Sipgate)
    angerufen werden kann ich jedoch nicht.

    Vieleicht kann mir jemand helfen

    Hier mein dialplan

    Code:
    [ Context 'sipgate' created by 'pbx_config' ]
      '800XXXX' =>      1. Dial(SIP/10|30|tr)                         [pbx_config]
                        2. Hangup()                                   [pbx_config]
    
    
    [ Context 'ausgsipgate' created by 'pbx_config' ]
      '_0.' =>          1. Dial(SIP/${EXTEN:1}@sipgate|30|tr)         [pbx_config]
                        2. Playback(invalid)                          [pbx_config]
                        3. Hangup()                                   [pbx_config]
    
    
    [ Context '11' created by 'pbx_config' ]
      '11' =>           1. Dial(SIP/11)                               [pbx_config]
                        2. Hangup()                                   [pbx_config]
    
    
    [ Context '10' created by 'pbx_config' ]
      '10' =>           1. Dial(SIP/10)                               [pbx_config]
                        2. ()                                         [pbx_config]
    
    
    [ Context 'default' created by 'pbx_config' ]
      Include =>        '10'                                          [pbx_config]
      Include =>        '11'                                          [pbx_config]
      Include =>        'ausgsipgate'                                 [pbx_config]
    
    [ Context 'parkedcalls' created by 'res_features' ]
      '700' =>          1. Park()                                     [res_features]
    
    und hier die konfigs

    sip.conf

    Code:
    
    [general]
    port=5060
    bindaddr=192.168.6.1
    context=default
    srvlookup=yes
    nat=yes
    insecure=very
    register=> 8006724:XXXXXX@sipgate.de/8006724
    
    
    [sipgate]
    type=friend
    username=8006724
    secret=XXXXXX
    host=sipgate.de
    fromuser=8006724
    nat=yes
    context=sipgate
    canreinvite=no
    
    
    [10]
    type=friend
    username=10
    secret=10
    host=dynamic
    callerid="10"=<10>
    
    
    [11]
    type=friend
    username=11
    secret=11
    host=dynamic
    callerid="11"=<11>
    
    extensions.conf

    Code:
    
    [general]
    static=yes
    writeprotect=no
    
    
    [default]
    include=> 10
    include=> 11
    include=> ausgsipgate
    
    
    
    [10]
    exten=>10,1,Dial(SIP/10)
    exten=>10,2 Hangup
    
    
    [11]
    exten=>11,1,Dial(SIP/11)
    exten=>11,2,Hangup
    
    [ausgsipgate]
    exten=> _0.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
    exten=> _0.,2,Playback(invalid)
    exten=> _0.,3,Hangup
    
    [sipgate]
    exten=> 800XXXX,1,Dial(SIP/10,30,tr)
    exten=> 800XXXX,2,Hangup
    
    
    hier meine peers

    Code:
    *CLI> sip show peers
    Name/username    Host            Dyn Nat ACL Mask             Port     Status
    11/11            192.168.6.5      D          255.255.255.255  5060     Unmonitored
    10/10            (Unspecified)    D          255.255.255.255  0        Unmonitored
    sipgate/8006724  217.10.79.9          N      255.255.255.255  5060     Unmonitored
    
    Und jetzt die debug ausgaben beim anruf
    Code:
    
    *CLI> sip debug
    SIP Debugging Enabled
    *CLI>
    
    Sip read:
    
    0 headers, 0 lines
    
    
    Sip read:
    INVITE sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 10:25:24 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 370
    Sipgate-Authentication: accepted
    
    v=0
    o=root 3372 3372 IN IP4 217.10.79.30
    s=session
    c=IN IP4 217.10.79.9
    t=0 0
    m=audio 47494 RTP/AVP 8 0 3 10 97 18 2 5
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=silenceSupp:off - - - -
    a=direction:active
    a=nortpproxy:yes
    
    18 headers, 17 lines
    Using latest request as basis request
    Sending to 217.10.79.9 : 5060 (NAT)
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 10
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 5
    Peer audio RTP is at port 217.10.79.9:47494
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format L16
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format DVI4
    Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
    Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
    Found peer 'sipgate'
    Reliably Transmitting (NAT):
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as4dd550ef
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:8006724@192.168.6.1>
    Proxy-Authenticate: Digest realm="asterisk", nonce="5b07b6b6"
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Scheduling destruction of call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30' in 15000 ms
    
    
    Sip read:
    INVITE sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 10:25:24 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 370
    Sipgate-Authentication: accepted
    
    v=0
    o=root 3372 3372 IN IP4 217.10.79.30
    s=session
    c=IN IP4 217.10.79.9
    t=0 0
    m=audio 47494 RTP/AVP 8 0 3 10 97 18 2 5
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=silenceSupp:off - - - -
    a=direction:active
    a=nortpproxy:yes
    
    18 headers, 17 lines
    Ignoring this request
    Found peer 'sipgate'
    
    
    Sip read:
    ACK sip:8006724@217.187.98.23:5060 SIP/2.0
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    To: <sip:4921158006724@sipgate.net>;tag=as4dd550ef
    CSeq: 102 ACK
    User-Agent: sipgate ser
    Content-Length: 0
    
    
    8 headers, 0 lines
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa668.cb0ddb76.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa668.ea40ee86.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK4b18c41f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 103 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=5
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa668.cb0ddb76.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa668.ea40ee86.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK4b18c41f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as4dd550ef
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 103 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:8006724@192.168.6.1>
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK7668.4995d0b3.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK7668.b33c6d73.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK5dfce868
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 104 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=8
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK7668.4995d0b3.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK7668.b33c6d73.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK5dfce868
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as1a4cc10a
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 104 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK8668.c8b81785.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK8668.e2ec3b54.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK37e7a872
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 105 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=0
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK8668.c8b81785.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK8668.e2ec3b54.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK37e7a872
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as73cea4b0
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 105 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3768.d837b7f7.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3768.afce1801.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6dc8e001
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 109 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=2
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3768.d837b7f7.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3768.afce1801.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6dc8e001
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as4d1f4c71
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 109 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=7
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as35a4a495
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKb478.6e124a07.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb478.8da83242.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK12f26619
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 110 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=4
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKb478.6e124a07.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb478.8da83242.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK12f26619
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as5add0b60
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 110 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=6
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as41d93c90
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=0
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as06db8db7
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=7
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as0097d44d
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=6
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as2b04d9bc
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as11b265de;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as11b265de;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=0
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as11b265de
    To: <sip:4921158006724@sipgate.net>;tag=as1b7a1d94
    Call-ID: 442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '442ed31d3e481cde2c1099cb590c3cfb@217.10.79.30'
    
    
    Sip read:
    
    0 headers, 0 lines
        -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
    11 headers, 0 lines
    Reliably Transmitting:
    REGISTER sip:sipgate.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK4dfd2398
    From: <sip:8006724@sipgate.de>;tag=as161f4ff1
    To: <sip:8006724@sipgate.de>
    Call-ID: 327b23c6643c98696633487374b0dc51@192.168.6.1
    CSeq: 106 REGISTER
    User-Agent: Asterisk PBX
    Expires: 120
    Contact: <sip:8006724@192.168.6.1>
    Event: registration
    Content-Length: 0
    
     (no NAT) to 217.10.79.9:5060
    
    
    Sip read:
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK4dfd2398;rport=5060;received=217.187.98.23
    From: <sip:8006724@sipgate.de>;tag=as161f4ff1
    To: <sip:8006724@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
    Call-ID: 327b23c6643c98696633487374b0dc51@192.168.6.1
    CSeq: 106 REGISTER
    WWW-Authenticate: Digest realm="sipgate.de", nonce="41b049624974c131304899dba5cc4de5f8a0c728"
    Server: sipgate ser
    Content-Length: 0
    Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8442 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1"
    
    
    10 headers, 0 lines
    12 headers, 0 lines
    Reliably Transmitting:
    REGISTER sip:sipgate.de SIP/2.0
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK47de180b
    From: <sip:8006724@sipgate.de>;tag=as161f4ff1
    To: <sip:8006724@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
    Call-ID: 327b23c6643c98696633487374b0dc51@192.168.6.1
    CSeq: 107 REGISTER
    User-Agent: Asterisk PBX
    Authorization: Digest username="8006724", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="41b049624974c131304899dba5cc4de5f8a0c728", response="634549faa5906b9c105ac86e6f2fbf88", opaque=""
    Expires: 120
    Contact: <sip:8006724@192.168.6.1>
    Event: registration
    Content-Length: 0
    
     (no NAT) to 217.10.79.9:5060
    
    
    Sip read:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK47de180b;rport=5060;received=217.187.98.23
    From: <sip:8006724@sipgate.de>;tag=as161f4ff1
    To: <sip:8006724@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
    Call-ID: 327b23c6643c98696633487374b0dc51@192.168.6.1
    CSeq: 107 REGISTER
    Contact: <sip:8006724@217.187.98.23:5060>;q=0.00;expires=120
    Server: sipgate ser
    Content-Length: 0
    Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8445 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1"
    
    
    10 headers, 0 lines
    Destroying call '327b23c6643c98696633487374b0dc51@192.168.6.1'
    
    
    Sip read:
    
    0 headers, 0 lines
     
    Die interne Nebenstelle kann ich aus irgendwelchen gründen auch nicht anwählen?

    Vieleicht kann mir ja jemand helfen
     
  2. Netview

    Netview IPPF-Promi

    Registriert seit:
    1 Apr. 2004
    Beiträge:
    3,366
    Zustimmungen:
    0
    Punkte für Erfolge:
    36
    Beruf:
    Dipl.-Inf.
    Ort:
    Westerwald
    Du willst eine pbx über sip callen?

    Dies geht nur über die capi z.B.

    exten => 8006724,1,Dial(Capi/@37:40,60)

    37 ist die msn der ISDN-Karte und 40 die Rufgruppe (gerufene Nummer an der Pbx)
     
  3. Hupe

    Hupe Aktives Mitglied

    Registriert seit:
    8 Apr. 2004
    Beiträge:
    2,586
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    10/10 (Unspecified) D 255.255.255.255 0 Unmonitored
    Also, eingehende Sipgate-Gespräche sollen wohl auf den Peer 10 weitergestelt werden, oder? Irgendwie sieht es nicht so aus, als ob der Client richtig registriert ist. Was ist das für ein Client. Setz doch mal "qualify=yes" bei den Peers dazu. Außerdem solltest Du mal codecs in der sip.conf definieren.
     
  4. BlackSektor

    BlackSektor Neuer User

    Registriert seit:
    3 Nov. 2004
    Beiträge:
    34
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Ein SNOM 190 und ein X-Lite Client (der ist nur zum Testen)
    Richtig, die Gespräche sollen an den Client 10 und 11 ran.
    Ich hab das ganze jetzt mal überarbeitet:

    Code:
    SIP Debugging Enabled for IP: 217.10.79.9:5060
    *CLI>
    
    Sip read:
    INVITE sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 16:33:44 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 372
    Sipgate-Authentication: accepted
    
    v=0
    o=root 24194 24194 IN IP4 217.10.79.30
    s=session
    c=IN IP4 217.10.79.9
    t=0 0
    m=audio 45596 RTP/AVP 8 0 3 10 97 18 2 5
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=silenceSupp:off - - - -
    a=direction:active
    a=nortpproxy:yes
    
    18 headers, 17 lines
    Using latest request as basis request
    Sending to 217.10.79.9 : 5060 (non-NAT)
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 10
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 5
    Peer audio RTP is at port 217.10.79.9:45596
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format L16
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format DVI4
    Capabilities: us - 0x80008(ALAW|H263), peer - audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), combined - 0x8(ALAW)
    Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
    Found peer 'sipgate'
    Reliably Transmitting (NAT):
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as4c4cb239
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:8006724@192.168.6.1>
    Proxy-Authenticate: Digest realm="asterisk", nonce="0a1b4fe7"
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Scheduling destruction of call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30' in 15000 ms
    
    
    Sip read:
    INVITE sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 16:33:44 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 372
    Sipgate-Authentication: accepted
    
    v=0
    o=root 24194 24194 IN IP4 217.10.79.30
    s=session
    c=IN IP4 217.10.79.9
    t=0 0
    m=audio 45596 RTP/AVP 8 0 3 10 97 18 2 5
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:10 L16/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:5 DVI4/8000
    a=silenceSupp:off - - - -
    a=direction:active
    a=nortpproxy:yes
    
    18 headers, 17 lines
    Ignoring this request
    Found peer 'sipgate'
    
    
    Sip read:
    ACK sip:8006724@217.187.98.23:5060 SIP/2.0
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    To: <sip:4921158006724@sipgate.net>;tag=as4c4cb239
    CSeq: 102 ACK
    User-Agent: sipgate ser
    Content-Length: 0
    
    
    8 headers, 0 lines
    11 headers, 0 lines
    Reliably Transmitting:
    OPTIONS sip:217.10.79.9 SIP/2.0
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138
    From: "asterisk" <sip:asterisk@192.168.6.1>;tag=as2d09910a
    To: <sip:217.10.79.9>
    Contact: <sip:asterisk@192.168.6.1>
    Call-ID: 01b7581e259646c21a44e3ca47596078@192.168.6.1
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 18:13:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Length: 0
    
     (no NAT) to 217.10.79.9:5060
    Retransmitting #1 (no NAT):
    OPTIONS sip:217.10.79.9 SIP/2.0
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138
    From: "asterisk" <sip:asterisk@192.168.6.1>;tag=as2d09910a
    To: <sip:217.10.79.9>
    Contact: <sip:asterisk@192.168.6.1>
    Call-ID: 01b7581e259646c21a44e3ca47596078@192.168.6.1
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX
    Date: Fri, 03 Dec 2004 18:13:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    
    
    Sip read:
    SIP/2.0 482 Loop Detected
    Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138;rport=5060;received=217.187.98.23
    From: "asterisk" <sip:asterisk@192.168.6.1>;tag=as2d09910a
    To: <sip:217.10.79.9>;tag=b11cb9bb270104b49a99a995b8c68544.1824
    Call-ID: 01b7581e259646c21a44e3ca47596078@192.168.6.1
    CSeq: 102 OPTIONS
    Server: sipgate ser
    Content-Length: 0
    Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8442 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:217.10.79.9 out_uri=sip:217.10.79.9 via_cnt==1"
    
    
    9 headers, 0 lines
    Destroying call '01b7581e259646c21a44e3ca47596078@192.168.6.1'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3e1.0ab748b7.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3e1.30ea5a67.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK36fccae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 103 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=5
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3e1.0ab748b7.1;received=217.10.79.9;rport=5060
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3e1.30ea5a67.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK36fccae9
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as4c4cb239
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 103 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:8006724@192.168.6.1>
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK0e1.c63cac2.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK0e1.678adc92.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK7d97ce0e
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 104 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=8
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK0e1.c63cac2.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK0e1.678adc92.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK7d97ce0e
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as093323a0
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 104 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK1e1.848513e5.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK1e1.36e770e6.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK1fe41bfd
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 105 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=0
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK1e1.848513e5.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK1e1.36e770e6.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK1fe41bfd
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as4af662a5
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 105 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKed1.968746b6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKed1.a82239f7.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0a9bb766
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=0
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKed1.968746b6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKed1.a82239f7.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0a9bb766
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as510dbbc9
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 106 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKfd1.335f65.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfd1.871d37b.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3e208b68
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=6
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKfd1.335f65.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfd1.871d37b.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3e208b68
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as3a181741
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 107 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKde1.726924c7.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKde1.9f80be33.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK228287fc
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=7
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKde1.726924c7.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKde1.9f80be33.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK228287fc
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as6280ba31
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 108 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKce1.4d28ba6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKce1.0d000fa5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3201dc42
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 109 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=2
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKce1.4d28ba6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKce1.0d000fa5.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3201dc42
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as1992b0db
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 109 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    
    Sip read:
    INFO sip:8006724@217.187.98.23:5060 SIP/2.0
    Record-Route: <sip:8006724@217.10.79.9;ftag=as36dc9bfc;lr=on>
    Max-Forwards: 9
    Record-Route: <sip:4921158006724@217.10.79.8;ftag=as36dc9bfc;lr=on>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4c2.9772d1e3.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4c2.78b08e04.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK66e9d372
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>
    Contact: <sip:08427985707@217.10.79.30>
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 110 INFO
    User-Agent: Asterisk PBX
    Content-Type: application/dtmf-relay
    Content-Length: 24
    Sipgate-Authentication: accepted
    
    Signal=4
    Duration=250
    
    16 headers, 2 lines
    Receiving DTMF!
    Transmitting (no NAT):
    SIP/2.0 481 Call leg/transaction does not exist
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4c2.9772d1e3.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4c2.78b08e04.0
    Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK66e9d372
    From: "08427985707" <sip:08427985707@217.10.79.30>;tag=as36dc9bfc
    To: <sip:4921158006724@sipgate.net>;tag=as3ae2bf16
    Call-ID: 7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30
    CSeq: 110 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact:
    Content-Length: 0
    
    
     to 217.10.79.9:5060
    Destroying call '7a2d479616a5c3ab4870f41f6fe9152d@217.10.79.30'
    
    Mir ist insbesondere das Aufgefallen
    SIP/2.0 481 Call leg / transaction does not exists

    Ports hab ich folgende Freigegeben (an den Asterisk
    5060, 10000 - 32000, und 21
    Fehlen da noch welche?

    Das ist meine Nebenstelle 11


    Code:
    *CLI> sip show peer 11
    
    
      * Name       : 11
      Secret       : <Set>
      MD5Secret    : <Not set>
      Context      : default
      Language     :
      FromUser     :
      FromDomain   :
      Callgroup    :  (0)
      Pickupgroup  :  (0)
      Mailbox      :
      LastMsgsSent : -1
      Dynamic      : Yes
      Expire       : 2
      Expiry       : 900
      Insecure     : No
      Nat          : No
      ACL          : No
      CanReinvite  : No
      PromiscRedir : No
      DTMFmode     : inband
      LastMsg      : 0
      ToHost       :
      Addr->IP     : 192.168.6.5 Port 5060
      Defaddr->IP  : 0.0.0.0 Port 5060
      Username     : 11
      Codecs       : ULAW ALAW
      Status       : OK (942 ms)
      Useragent    :
      Full Contact : sip:11@192.168.6.5:5060;line=pxtx8eem
    
    Und das die Verbindung zu sipgate

    Code:
    *CLI> sip show peer sipgate
    
    
      * Name       : sipgate
      Secret       : <Set>
      MD5Secret    : <Not set>
      Context      : sipgate
      Language     :
      FromUser     : 8006724
      FromDomain   :
      Callgroup    :  (0)
      Pickupgroup  :  (0)
      Mailbox      :
      LastMsgsSent : -1
      Dynamic      : No
      Expire       : -1
      Expiry       : 900
      Insecure     : No
      Nat          : Always
      ACL          : No
      CanReinvite  : No
      PromiscRedir : No
      DTMFmode     : rfc2833
      LastMsg      : 0
      ToHost       : sipgate.de
      Addr->IP     : 217.10.79.9 Port 5060
      Defaddr->IP  : 0.0.0.0 Port 0
      Username     : 8006724
      Codecs       : ALAW H.263
      Status       : OK (99 ms)
      Useragent    :
      Full Contact :
    
    Alle Nebenstellen

    Code:
    *CLI> sip show peers
    Name/username    Host            Dyn Nat ACL Mask             Port     Status
    11/11            192.168.6.5      D          255.255.255.255  5060     OK (43 ms)
    10/10            (Unspecified)    D          255.255.255.255  0        Unmonitored
    sipgate/8006724  217.10.79.9          N      255.255.255.255  5060     OK (97 ms)
    
    Meine überarbeitete sip.conf
    Code:
    
    [general]
    port=5060
    bindaddr=192.168.6.1
    context=default
    srvlookup=yes
    insecure=very
    disallow=gsm
    allow=alaw
    disallow=ulaw
    register=> 8006724:******@sipgate.de/8006724
    
    
    [sipgate]
    type=friend
    username=8006724
    secret=*******               
    host=sipgate.de
    fromuser=8006724
    nat=yes
    context=sipgate
    canreinvite=no
    qualify=yes
    
    [10]
    type=friend
    username=10
    secret=10
    host=dynamic
    callerid="10"=<10>
    nat=no
    
    [11]
    type=friend
    username=11
    secret=11
    host=dynamic
    callerid="11"=<11>
    dtmfmode=inband
    disallow=all
    allow=ulaw
    allow=alaw
    qualify=yes
    canreinvite=no
    nat=no
    
    Die extensions.conf

    Code:
    
    
    [general]
    static=yes
    writeprotect=no
    
    
    [default]
    include=> 10
    include=> 11
    include=> ausgsipgate
    
    
    
    [10]
    exten=>10,1,Dial(SIP/10)
    exten=>10,2 Hangup
    
    
    [11]
    exten=>11,1,Dial(SIP/11)
    exten=>11,2,Hangup
    
    [ausgsipgate]
    exten=> _0.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
    exten=> _0.,2,Playback(invalid)
    exten=> _0.,3,Hangup
    
    [sipgate]
    exten=> h,1,Hangup
    exten=> 800XXXX,1,Dial(SIP/11,30,tr)
    
    Ich weis echt nicht mehr weiter, Ich hock wirklich schon seit heute vormittag am Problem rum und weis nicht was falsch ist.
     
  5. BlackSektor

    BlackSektor Neuer User

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    Mein Router ist UPNP fähig. Deshalb hab ich jetzt mal das ganze Portforwarding rausgemacht.
    Es funktioniert immer noch nicht.
     
  6. Hupe

    Hupe Aktives Mitglied

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    Hmm, das tut mir echt leid, dass es nicht funzt. Ein paar kleinigkeiten habe ich noch:
    ergänze mal in der sip conf:
    bei [sipgate]
    insecure=very
    fromdomain=sipgate.de
    disallow=all
    allow=g726
    allow=alaw
    allow=ulaw

    außerdem sollte bei [general]:
    disallow=all und nicht disallow=gsm stehen.
    Sonst ist alles außer GSM erlaubt.
     
  7. BlackSektor

    BlackSektor Neuer User

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    Hallo, jetzt komm ich gar nicht mehr raus.

    Könnte mir mal jemand, der sipgate erfolgrich mit Nebenstellen am laufen hat seine sip.conf und seinen dialplan zuschicken.

    Außdem wäre es mal interessant welche Ports ich am Netgear WGR 614 v. 4 (neueste Firmware) freigeben muß. Bei UPNP hätte ich gedacht, daß ich gar keine freigeben muß.

    Ich hab auch schon versucht, die DMZ auf dem Asterisk zu legen. Funktioniert auch nicht.



    Bitte die confs per PN oder Board-E-Mail bzw. Info, dann gebe ich die richtige Adresse.

    Danke schonmal.

    Edit von Christoph: Habe die E-Mail-Adresse zum Schutz vor Spam entfernt.


    OK, bitte konfigs am mein Board Postfach Danke!
     
  8. BlackSektor

    BlackSektor Neuer User

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    Keiner da, der mir seine Konfigs per PN zukommen lassen kann (Natürlich ohne Passwort)
     
  9. Netview

    Netview IPPF-Promi

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    damit klingeln dann beide Geräte bei eingehenden sipgate- Rufen:

    in der extensions.conf:

    exten => 8006724,1,Dial(SIP/10&SIP/11,60)
     
  10. BlackSektor

    BlackSektor Neuer User

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    So, raus bzw. reintelefonieren geht. Blos ich höre den anderen gesprächspartner nicht und es wird jeweils aufgelegt. Meine configs stimmen soweit (denke ich). Mein * befindet sich hinter dem DSL Router.
    Giebts den Router, die das korrekt durchleiten können, oder muß ich wirklich den * Server auch zum DSL Router machen?
     
  11. Netview

    Netview IPPF-Promi

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    Folgende ports sind freizugeben:

    10000-20000 (udp) gemäss rtp.conf
    5060 (udp) gemäss sip.conf (port=5060 unter general:)
    NAT=yes setzen (für alle clients und peers)!