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Stummer Anrufer die Zweite :(

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von barmeier, 21 Feb. 2006.

  1. barmeier

    barmeier Neuer User

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    Hallo,

    dank des Forums habe ich meinen Asterisk mit C2 AVM und Anlagenanschluss
    zum Laufen gebracht alles läuft Bestens... bis auf folgendes Problem:

    Wenn ich interne Gespräche führe, also SIP->SIP kann der Anrufer vom
    angerufenen nicht gehört werden. Drückt einer der beiden zweimal nacheinander
    auf die HOLD Taste des Telefons können sich beide hören. Das passiert häufig,
    aber nicht immer und ist absoluter Terror, wenn man ein Gespräch von Außen
    vermittelt bekommt.

    Ich habe keine Firewalls im internen Netz und ich habe keine Protokolle auf einem der Switches oder Router geblockt.

    Was mich wundert, ist das im Telefon Setup als RTP Port 1722 angegeben wird, in der rtp.conf aber

    rtpstart=10000
    rtpend=20000

    steht. WIe funktioniert das überhaupt ??

    Hat jemand eine Idee wie ich mein Problem lösen kann ??

    Danke im voraus.

    Ciao
    Matze
     
  2. madiehl

    madiehl Mitglied

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    Hast Du denn die Ports 10000 bis 20000 UDP an den Asterisk weitergeleitet?
     
  3. barmeier

    barmeier Neuer User

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    Hi madiehl,

    was genau verstehst du unter weitergeleitet ?
    Telefone und Asterisk sind im internen Netz, keine Firewall, kein NAT, kein Portforwarding.

    Was ich nicht verstehe ist, wenn auf dem Telefon RTP Port 1722 angegeben wird, wie
    kann der Asterisk überhaupt antworten ??

    Nimmt er jeden Datenverkehr per UDP auf, der das richtige Format hat ???

    Ich werde weiter probieren, aber vielleicht hast du oder jemand anderes noch einen Hinweis
    für mich.

    Ciao
    Matze
     
  4. barmeier

    barmeier Neuer User

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    Hallo zusammen,

    ich konnte einen Workaround erarbeiten. Wie es aussieht reicht es ein SIP DEBUG auf der Konsole einzugeben. Solange der Debugmodus aktiv ist läuft alles perfekt.

    Sobald ich SIP NO DEBUG eingebe geht es nicht mehr :confused: :confused:

    Hat jemand so etwas schon mal gesehen und kann mir erklären was das bedeutet ??

    Ciao
    Matze
     
  5. chaos2000

    chaos2000 Aktives Mitglied

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    Ort:
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    ist für das telefon nat=no gesetzt?
    sonst kannst du ja mal das debug posten
     
  6. barmeier

    barmeier Neuer User

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    Hi,

    ich habe nat=no und nat=never versucht. Hier ist meine sip.conf

    Code:
    [general]
    context=default                 ; Default context for incoming calls
    bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
    bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
    callevents=yes
    ;srvlookup=yes                  ; Enable DNS SRV lookups on outbound calls
                                    ; Note: Asterisk only uses the first host
                                    ; in SRV records
                                    ; Disabling DNS SRV lookups disables the
                                    ; ability to place SIP calls based on domain
                                    ; names to some other SIP users on the Internet
    
    nat=never
    tos=184
    
    disallow=all                    ; First disallow all codecs
    allow=ulaw
    allow=alaw                      ; Allow codecs in order of preference
    allow=gsm
    allow=g723
    allow=g729
    musicclass=default
    
    [12]
    type=friend
    host=192.168.10.12
    username=12
    authname=12
    secret=1234
    context=default
    
    [13]
    type=friend
    host=192.168.10.13
    callerid=398068313
    username=13
    authname=13
    secret=1234
    context=default
    
    [14]
    type=friend
    host=192.168.10.14
    username=14
    authname=14
    secret=1234
    context=default
    
    [15]
    type=friend
    host=192.168.10.15
    username=15
    authname=15
    context=default
    
    [16]
    type=friend
    ;host=192.168.10.16
    host=dynamic
    username=16
    authname=16
    secret=1234
    context=default
    
    [20]
    type=friend
    ;host=dynamic
    ;fromuser=398068320
    host=192.168.10.20
    username=20
    authname=20
    secret=1234
    context=default
    
    
    Ein SIP Debug kann ich nur erzeugen, in dem ich es ein paar mal an und abschalte (das SIP Debug, dann tritt das Problem auch mit aktiviertem Debug auf :confused:

    Das Log enthält den Versuch einer Verbindung, in der der Anrufer keinen Audio vom angerufenen bekommen hat, der Angerufene aber alles wunder bar hören konnte.

    Code:
    <-- SIP read from 192.168.10.16:5060:
    INVITE sip:20@192.168.10.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp
    Max-Forwards: 70
    To: <sip:20@192.168.10.1>
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 850 INVITE
    Contact: <sip:16.192.168.10.1@192.168.10.16>
    Content-Type: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
    Supported: 100rel
    User-Agent: Twinkle/0.5
    Content-Length: 250
    
    v=0
    o=16 1711186251 423106286 IN IP4 192.168.10.16
    s=-
    c=IN IP4 192.168.10.16
    t=0 0
    m=audio 8000 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    
    --- (13 headers 12 lines)---
    Using INVITE request as basis request - vaqfohjoalascft@192.168.10.16
    Sending to 192.168.10.16 : 5060 (non-NAT)
    Reliably Transmitting (no NAT) to 192.168.10.16:5060:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp;received=192.168.10.16
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    To: <sip:20@192.168.10.1>;tag=as4531458b
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 850 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Max-Forwards: 70
    Contact: <sip:20@192.168.10.1>
    Proxy-Authenticate: Digest realm="asterisk", nonce="1f61e569"
    Content-Length: 0
    
    
    ---
    Scheduling destruction of call 'vaqfohjoalascft@192.168.10.16' in 15000 ms
    Found user '16'
    
    <-- SIP read from 192.168.10.16:5060:
    ACK sip:20@192.168.10.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKckbkvbbp
    Max-Forwards: 70
    To: <sip:20@192.168.10.1>;tag=as4531458b
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 850 ACK
    User-Agent: Twinkle/0.5
    Content-Length: 0
    
    
    --- (9 headers 0 lines)---
    
    <-- SIP read from 192.168.10.16:5060:
    INVITE sip:20@192.168.10.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh
    Max-Forwards: 70
    Proxy-Authorization: Digest username="16",realm="asterisk",nonce="1f61e569",uri="sip:20@192.168.10.1",response="08649ec6931be6e927f7c8023dc591b5",algorithm=MD5
    To: <sip:20@192.168.10.1>
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 851 INVITE
    Contact: <sip:16.192.168.10.1@192.168.10.16>
    Content-Type: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
    Supported: 100rel
    User-Agent: Twinkle/0.5
    Content-Length: 250
    
    v=0
    o=16 1711186251 423106286 IN IP4 192.168.10.16
    s=-
    c=IN IP4 192.168.10.16
    t=0 0
    m=audio 8000 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    
    --- (14 headers 12 lines)---
    Using INVITE request as basis request - vaqfohjoalascft@192.168.10.16
    Sending to 192.168.10.16 : 5060 (non-NAT)
    Found user '16'
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 3
    Found RTP audio format 101
    Peer audio RTP is at port 192.168.10.16:8000
    Found description format PCMU
    Found description format PCMA
    Found description format GSM
    Found description format telephone-event
    Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Looking for 20 in default (domain 192.168.10.1)
    list_route: hop: <sip:16.192.168.10.1@192.168.10.16>
    Transmitting (no NAT) to 192.168.10.16:5060:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    To: <sip:20@192.168.10.1>
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 851 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Max-Forwards: 70
    Contact: <sip:20@192.168.10.1>
    Content-Length: 0
    
    
    ---
    We're at 192.168.10.1 port 18746
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x1 (g723) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    13 headers, 15 lines
    Reliably Transmitting (no NAT) to 192.168.10.20:5060:
    INVITE sip:20@192.168.10.20 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Wed, 22 Feb 2006 15:46:18 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 332
    
    v=0
    o=root 5499 5499 IN IP4 192.168.10.1
    s=session
    c=IN IP4 192.168.10.1
    t=0 0
    m=audio 18746 RTP/AVP 0 8 3 4 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
    
    <-- SIP read from 192.168.10.20:5060:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 102 INVITE
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:20@192.168.10.20:5060>
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    Transmitting (no NAT) to 192.168.10.16:5060:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    To: <sip:20@192.168.10.1>;tag=as3f36a918
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 851 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Max-Forwards: 70
    Contact: <sip:20@192.168.10.1>
    Content-Length: 0
    
    
    ---
    
    <-- SIP read from 192.168.10.20:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK63919bb3
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 102 INVITE
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:20@192.168.10.20:5060>
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 139
    
    v=0
    o=- 91792934 19784135 IN IP4 192.168.10.20
    s=SIP CALL
    c=IN IP4 192.168.10.20
    t=0 0
    m=audio 17220 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    
    --- (11 headers 7 lines)---
    Found RTP audio format 0
    Peer audio RTP is at port 192.168.10.20:17220
    Found description format PCMU
    Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    list_route: hop: <sip:20@192.168.10.20:5060>
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    Transmitting (no NAT) to 192.168.10.20:5060:
    ACK sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK13254a83
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    We're at 192.168.10.1 port 11514
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x1 (g723) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 192.168.10.16:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKyqyssgnh;received=192.168.10.16
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    To: <sip:20@192.168.10.1>;tag=as3f36a918
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 851 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Max-Forwards: 70
    Contact: <sip:20@192.168.10.1>
    Content-Type: application/sdp
    Content-Length: 332
    
    v=0
    o=root 5499 5499 IN IP4 192.168.10.1
    s=session
    c=IN IP4 192.168.10.1
    t=0 0
    m=audio 11514 RTP/AVP 0 8 3 4 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    We're at 192.168.10.1 port 18746
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    13 headers, 10 lines
    Reliably Transmitting (no NAT) to 192.168.10.20:5060:
    INVITE sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33e07126
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    X-asterisk-info: SIP re-invite (RTP bridge)
    Content-Type: application/sdp
    Content-Length: 206
    
    v=0
    o=root 5499 5500 IN IP4 192.168.10.16
    s=session
    c=IN IP4 192.168.10.16
    t=0 0
    m=audio 8000 RTP/AVP 0 8 3
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=silenceSupp:off - - - -
    
    ---
    
    <-- SIP read from 192.168.10.16:5060:
    ACK sip:20@192.168.10.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKnlokqyna
    Max-Forwards: 70
    Proxy-Authorization: Digest username="16",realm="asterisk",nonce="1f61e569",uri="sip:20@192.168.10.1",response="08649ec6931be6e927f7c8023dc591b5",algorithm=MD5
    To: <sip:20@192.168.10.1>;tag=as3f36a918
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 851 ACK
    User-Agent: Twinkle/0.5
    Content-Length: 0
    
    
    --- (10 headers 0 lines)---
    set_destination: Parsing <sip:16.192.168.10.1@192.168.10.16> for address/port to send to
    set_destination: set destination to 192.168.10.16, port 5060
    We're at 192.168.10.1 port 11514
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    13 headers, 10 lines
    Reliably Transmitting (no NAT) to 192.168.10.16:5060:
    INVITE sip:16.192.168.10.1@192.168.10.16 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK757884bd
    From: <sip:20@192.168.10.1>;tag=as3f36a918
    To: "16" <sip:16@192.168.10.1>;tag=nhzez
    Contact: <sip:20@192.168.10.1>
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    X-asterisk-info: SIP re-invite (RTP bridge)
    Content-Type: application/sdp
    Content-Length: 216
    
    v=0
    o=root 5499 5500 IN IP4 192.168.10.20
    s=session
    c=IN IP4 192.168.10.20
    t=0 0
    m=audio 17220 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
    
    <-- SIP read from 192.168.10.20:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33e07126
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 103 INVITE
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:20@192.168.10.20:5060>
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 139
    
    v=0
    o=- 99407365 84190217 IN IP4 192.168.10.20
    s=SIP CALL
    c=IN IP4 192.168.10.20
    t=0 0
    m=audio 17220 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    
    --- (11 headers 7 lines)---
    Found RTP audio format 0
    Peer audio RTP is at port 192.168.10.20:17220
    Found description format PCMU
    Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    Transmitting (no NAT) to 192.168.10.20:5060:
    ACK sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3458c279
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 103 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    
    <-- SIP read from 192.168.10.16:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK757884bd
    To: "16" <sip:16@192.168.10.1>;tag=nhzez
    From: <sip:20@192.168.10.1>;tag=as3f36a918
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 102 INVITE
    Contact: <sip:16.192.168.10.1@192.168.10.16>
    Content-Type: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE
    Server: Twinkle/0.5
    Supported:
    Content-Length: 203
    
    v=0
    o=16 1711186251 423106286 IN IP4 192.168.10.16
    s=-
    c=IN IP4 192.168.10.16
    t=0 0
    m=audio 8000 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    
    --- (12 headers 10 lines)---
    Found RTP audio format 0
    Found RTP audio format 101
    Peer audio RTP is at port 192.168.10.16:8000
    Found description format PCMU
    Found description format telephone-event
    Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    list_route: hop: <sip:16.192.168.10.1@192.168.10.16>
    set_destination: Parsing <sip:16.192.168.10.1@192.168.10.16> for address/port to send to
    set_destination: set destination to 192.168.10.16, port 5060
    Transmitting (no NAT) to 192.168.10.16:5060:
    ACK sip:16.192.168.10.1@192.168.10.16 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK7b2fc0cf
    From: <sip:20@192.168.10.1>;tag=as3f36a918
    To: "16" <sip:16@192.168.10.1>;tag=nhzez
    Contact: <sip:20@192.168.10.1>
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    
    <-- SIP read from 192.168.10.16:5060:
    BYE sip:20@192.168.10.1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKrrqpuppk
    Max-Forwards: 70
    To: <sip:20@192.168.10.1>;tag=as3f36a918
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 852 BYE
    User-Agent: Twinkle/0.5
    Content-Length: 0
    
    
    --- (9 headers 0 lines)---
    Sending to 192.168.10.16 : 5060 (non-NAT)
    Transmitting (no NAT) to 192.168.10.16:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.16;rport;branch=z9hG4bKrrqpuppk;received=192.168.10.16
    From: "16" <sip:16@192.168.10.1>;tag=nhzez
    To: <sip:20@192.168.10.1>;tag=as3f36a918
    Call-ID: vaqfohjoalascft@192.168.10.16
    CSeq: 852 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Max-Forwards: 70
    Contact: <sip:20@192.168.10.1>
    Content-Length: 0
    X-Asterisk-HangupCause: Normal Clearing
    
    
    ---
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    We're at 192.168.10.1 port 18746
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x1 (g723) to SDP
    Adding codec 0x100 (g729) to SDP
    13 headers, 13 lines
    Reliably Transmitting (no NAT) to 192.168.10.20:5060:
    INVITE sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK1d9fdd72
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 104 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    X-asterisk-info: SIP re-invite (RTP bridge)
    Content-Type: application/sdp
    Content-Length: 276
    
    v=0
    o=root 5499 5501 IN IP4 192.168.10.1
    s=session
    c=IN IP4 192.168.10.1
    t=0 0
    m=audio 18746 RTP/AVP 0 8 3 4 18
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=silenceSupp:off - - - -
    
    ---
    
    <-- SIP read from 192.168.10.20:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK1d9fdd72
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 104 INVITE
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:20@192.168.10.20:5060>
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 139
    
    v=0
    o=- 81192499 69731988 IN IP4 192.168.10.20
    s=SIP CALL
    c=IN IP4 192.168.10.20
    t=0 0
    m=audio 17220 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    
    --- (11 headers 7 lines)---
    Found RTP audio format 0
    Peer audio RTP is at port 192.168.10.20:17220
    Found description format PCMU
    Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    Transmitting (no NAT) to 192.168.10.20:5060:
    ACK sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3683ba3e
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 104 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    set_destination: Parsing <sip:20@192.168.10.20:5060> for address/port to send to
    set_destination: set destination to 192.168.10.20, port 5060
    Reliably Transmitting (no NAT) to 192.168.10.20:5060:
    BYE sip:20@192.168.10.20:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK2906a3f4
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:16@192.168.10.1>
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 105 BYE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    Destroying call 'vaqfohjoalascft@192.168.10.16'
    
    <-- SIP read from 192.168.10.20:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK2906a3f4
    Call-ID: 129bb3ae5aa75c442d3e887577154d87@192.168.10.1
    CSeq: 105 BYE
    From: "16" <sip:16@192.168.10.1>;tag=as04d6e020
    To: <sip:20@192.168.10.20>;tag=IPfP7FSeAp07cigB
    Contact: <sip:20@192.168.10.20:5060>
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    Destroying call '129bb3ae5aa75c442d3e887577154d87@192.168.10.1'
    
    
    
     
  7. barmeier

    barmeier Neuer User

    Registriert seit:
    9 Feb. 2006
    Beiträge:
    58
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Hi zusammen,

    ich habe das jetzt so gelöst, dass ich meine Telefone mit IAX2 Images bestückt habe und siehe da schon gehts.

    Warum das vorher nicht ging weiss ich allerdings immer noch nicht :(

    Vielen Dank.

    Ciao
    Matze