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Keine ausgehende Anrufe von Freenet

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von Groper, 13 Juni 2005.

  1. Groper

    Groper Neuer User

    Registriert seit:
    12 Juni 2005
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    Hallo Forum,

    Ich kann nicht von Freenet nach außen anrufen. Gleichzeitig nutze ich zwei SIP Provider: Sipgate und Freenet. Mit Sipgate kann ich beiderseitig anrufen, und bei Freenet nur ankommende Anrufe kommen. * steht unter NAT. An DSL Modem ZyXel Prestige 660HW-67 habe ich Ports 5060(TCP und UDP) und 10000-20000(UDP) zu * weitergeleitet. In interne Netz (192.168.1.0/255.255.255.0) nutze ich X-Lite als Client *.


    Hier sind meine Konfigurationsdateien:

    Sip.conf:
    Code:
    [general] 
    port=5060 
    bindaddr=0.0.0.0
    nat=yes
    externip=xxx.dyndns.org
    ;externhost=xxx.dyndns.org
    localnet=192.168.1.0/255.255.255.0
    context=default
    tos=reliability
    qualify=yes
    srvlookup=yes
    disallow=all
    allow=alaw
    allow=ulaw 
    allow=gsm
    insecure=very
    language=de
    canreinvite=no
    dtmfmode=info
    maxexpirey=3600
    defaultexpirey=600
    
    
    register => username:*******@sipgate.de/username
    register => username:*******@freenet.de/username
    
    
    [freenet]
    type=friend
    username= username
    host=freenet.de
    secret=********
    fromuser=username
    fromdomain=freenet.de
    canreinvite=no
    nat=yes
    qualify=no
    insecure=very
    disallow=all
    allow=alaw
    allow=ulaw 
    allow=gsm      
    dtmfmode=info
    context=default
    
    [sipgate] 
    type=friend 
    username=username
    host=sipgate.de
    secret=****** 
    fromuser=username
    fromdomain=sipgate.de 
    canreinvite=no 
    nat=no
    qualify=no 
    insecure=very
    dtmfmode=info
    
    
    [10] 
    type=friend 
    username=10
    secret=10
    host=dynamic 
    qualify=1200 
    nat=yes   
    canreinvite=no
    
    [11] 
    type=friend 
    username=11 
    secret=11 
    host=dynamic 
    ;qualify=1200
    nat=yes
    canreinvite=no
    
    [12] 
    type=friend 
    username=12 
    secret=12
    host=dynamic 
    ;qualify=1200
    nat=yes
    canreinvite=no
    
    extensions.conf:
    Code:
    [general]
    static=yes
    writeprotect=yes
    
    [default] 
    include => 10 
    include => 11
    include => 12 
    include => fromsip 
    include => tosip 
         
    [10] 
    exten => 10,1,Dial(SIP/10,60) 
    exten => 10,2,Hangup 
        
    [11]
    exten => 11,1,Dial(SIP/11,45)
    exten => 11,2,Hangup
    
    [12] 
    exten => 12,1,Dial(SIP/12,10)
    exten => 12,2,Hangup 
    
    [13] 
    exten => 13,1,Dial(SIP/12,45) 
    exten => 13,2,Hangup 
    	    
    [tosip] 
    exten => _9.,1,Dial(SIP/${EXTEN:1}@freenet,60,tr) 
    exten => _9.,2,Hangup
    exten => _1.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr) 
    exten => _1.,2,Hangup 
    		  
    [fromsip] 
    ;freenet
    exten => username,1,Dial(SIP/12,20,tr) 
    exten => username,2,Hangup
    ;sipgate
    exten => username,1,Dial(SIP/12,20,tr)
    extem => username,2,Hangup
    
    Wenn ich ausgehende Anruf von Freenet versuche zu machen, krige ich CLI Meldung:
    Code:
        -- Executing Dial("SIP/12-9c93", "SIP/040123456@freenet|60|tr") in new stack
            -- Called 040123456@freenet
    	Jun 12 20:31:40 WARNING[1108048816]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 5b28dc0b179c16020c79c4457b8382fb@84.140.46.80 for seqno 102 (Critical Request)
    	  == No one is available to answer at this time
    	      -- Executing Hangup("SIP/12-9c93", "") in new stack
    	        == Spawn extension (default, 9040123456, 2) exited non-zero on 'SIP/12-9c93'
    		Jun 12 20:31:46 WARNING[1108048816]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 5b28dc0b179c16020c79c4457b8382fb@84.140.46.80 for seqno 102 (Non-critical Request)
    		
    
    Mit Befehl "sip debug peer ...." krige ich Meldungen:

    sip debug sipgate :)
    Code:
    sip debug peer sipgate   
    SIP Debugging Enabled for IP: 217.10.79.9:5060
        -- Executing Dial("SIP/12-66b9", "SIP/040123456@sipgate|30|tr") in new stack
        We're at 192.168.1.3 port 14010
        Answering/Requesting with root capability 4
        Answering with capability 0x2(GSM)
        Answering with capability 0x8(ALAW)
        12 headers, 10 lines
        Reliably Transmitting:
        INVITE sip:040123456@sipgate.de SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3431b5a4
        From: "12" <sip:030123456@sipgate.de>;tag=as06a34a3d
        To: <sip:040123456@sipgate.de>
        Contact: <sip:030123456@192.168.1.3>
        Call-ID: 08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Date: Sat, 11 Jun 2005 20:26:25 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
        Content-Type: application/sdp
        Content-Length: 203
        
        v=0
        o=root 4347 4347 IN IP4 192.168.1.3
        s=session
        c=IN IP4 192.168.1.3
        t=0 0
        m=audio 14010 RTP/AVP 0 3 8
        a=rtpmap:0 PCMU/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:8 PCMA/8000
        a=silenceSupp:off - - - -
         (no NAT) to 217.10.79.9:5060
             -- Called 040123456@sipgate
    	 Transmitting:
    	 ACK sip:040123456@sipgate.de SIP/2.0
    	 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3431b5a4
    	 From: "12" <sip:030123456@sipgate.de>;tag=as06a34a3d
    	 To: <sip:040123456@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.23ed
    	 Contact: <sip:030123456@192.168.1.3>
    	 Call-ID: 08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3
    	 CSeq: 102 ACK
    	 User-Agent: Asterisk PBX
    	 Content-Length: 0
    	 
    	  (no NAT) to 217.10.79.9:5060
    	  We're at 192.168.1.3 port 14010
    	  Answering/Requesting with root capability 4
    	  Answering with capability 0x2(GSM)
    	  Answering with capability 0x8(ALAW)
    	  Reliably Transmitting:
    	  INVITE sip:040123456@sipgate.de SIP/2.0
    	  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK6a8bcaf0
    	  From: "12" <sip:030123456@sipgate.de>;tag=as06a34a3d
    	  To: <sip:040123456@sipgate.de>
    	  Contact: <sip:030123456@192.168.1.3>
    	  Call-ID: 08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3
    	  CSeq: 103 INVITE
    	  User-Agent: Asterisk PBX
    	  Proxy-Authorization: Digest username="030123456", realm="sipgate.de", algorithm=MD5, uri="sip:040123456@sipgate.de", nonce="42ab4936ecb5d26c5735a606d6d6800d172b00d5", response="7f69eb2f6359f31e9190ddebaf2d2127", opaque=""
    	  Date: Sat, 11 Jun 2005 20:26:26 GMT
    	  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	  Content-Type: application/sdp
    	  Content-Length: 203
    	  
    	  v=0
    	  o=root 4347 4348 IN IP4 192.168.1.3
    	  s=session
    	  c=IN IP4 192.168.1.3
    	  t=0 0
    	  m=audio 14010 RTP/AVP 0 3 8
    	  a=rtpmap:0 PCMU/8000
    	  a=rtpmap:3 GSM/8000
    	  a=rtpmap:8 PCMA/8000
    	  a=silenceSupp:off - - - -
    	   (no NAT) to 217.10.79.9:5060
    	   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 217.10.67.3:10054
    	   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)
    	   list_route: hop: <sip:040123456@217.10.79.9;ftag=as06a34a3d;lr=on>
    	   list_route: hop: <sip:4940123456@217.10.79.8;ftag=as06a34a3d;lr=on>
    	   list_route: hop: <sip:4940123456@217.10.67.3>
    	   set_destination: Parsing <sip:040123456@217.10.79.9;ftag=as06a34a3d;lr=on> for address/port to send to
    	   set_destination: set destination to 217.10.79.9, port 5060
    	   Transmitting:
    	   ACK sip:040123456@sipgate.de SIP/2.0
    	   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK00584e38
    	   Route: <sip:4940123456@217.10.79.8;ftag=as06a34a3d;lr=on>,<sip:4940123456@217.10.67.3>
    	   From: "12" <sip:030123456@sipgate.de>;tag=as06a34a3d
    	   To: <sip:040123456@sipgate.de>;tag=as7632f45f
    	   Contact: <sip:030123456@192.168.1.3>
    	   Call-ID: 08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3
    	   CSeq: 103 ACK
    	   User-Agent: Asterisk PBX
    	   Content-Length: 0
    	   
    	    (no NAT) to 217.10.79.9:5060
    	        -- SIP/sipgate-be97 answered SIP/12-66b9
    		    -- Attempting native bridge of SIP/12-66b9 and SIP/sipgate-be97
    		    Sending to 217.10.79.9 : 5060 (non-NAT)
    		    Transmitting (no NAT):
    		    SIP/2.0 200 OK
    		    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5219.4041cb6.0
    		    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5219.8fc88f12.0
    		    Via: SIP/2.0/UDP 217.10.67.3:5060;branch=z9hG4bK489adfbf
    		    Record-Route: <sip:030123456@217.10.79.9;ftag=as7632f45f;lr=on>
    		    Record-Route: <sip:030123456@217.10.79.8;ftag=as7632f45f;lr=on>
    		    From: <sip:040123456@sipgate.de>;tag=as7632f45f
    		    To: "12" <sip:030123456@sipgate.de>;tag=as06a34a3d
    		    Call-ID: 08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3
    		    CSeq: 102 BYE
    		    User-Agent: Asterisk PBX
    		    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    		    Contact: <sip:030123456@192.168.1.3>
    		    Content-Length: 0
    		    
    		    
    		     to 217.10.79.9:5060
    		       == Spawn extension (default, 1040123456, 1) exited non-zero on 'SIP/12-66b9'
    		       Destroying call '08a106f530bfe7be0e02ca7b70f411b4@192.168.1.3'
    		       
    		       
    
    und sip debug freenet :(
    Code:
    *CLI> sip debug peer freenet
    SIP Debugging Enabled for IP: 62.104.23.42:5060
        -- Executing Dial("SIP/12-7374", "SIP/040123456@freenet|60|tr") in new stack
        We're at 192.168.1.3 port 11340
        Answering/Requesting with root capability 4
        Answering with capability 0x2(GSM)
        Answering with capability 0x8(ALAW)
        12 headers, 10 lines
        Reliably Transmitting:
        INVITE sip:040123456@freenet.de SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
        From: "12" <sip:username@freenet.de>;tag=as3c2deb15
        To: <sip:040123456@freenet.de>
        Contact: <sip:username@192.168.1.3>
        Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Date: Sat, 11 Jun 2005 19:45:26 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
        Content-Type: application/sdp
        Content-Length: 203
        
        v=0
        o=root 4282 4282 IN IP4 192.168.1.3
        s=session
        c=IN IP4 192.168.1.3
        t=0 0
        m=audio 11340 RTP/AVP 0 3 8
        a=rtpmap:0 PCMU/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:8 PCMA/8000
        a=silenceSupp:off - - - -
         (no NAT) to 62.104.23.42:5060
             -- Called 040123456@freenet
    	 Retransmitting #1 (no NAT):
    	 INVITE sip:040123456@freenet.de SIP/2.0
    	 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	 From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	 To: <sip:040123456@freenet.de>
    	 Contact: <sip:username@192.168.1.3>
    	 Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	 CSeq: 102 INVITE
    	 User-Agent: Asterisk PBX
    	 Date: Sat, 11 Jun 2005 19:45:26 GMT
    	 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	 Content-Type: application/sdp
    	 Content-Length: 203
    	 
    	 v=0
    	 o=root 4282 4282 IN IP4 192.168.1.3
    	 s=session
    	 c=IN IP4 192.168.1.3
    	 t=0 0
    	 m=audio 11340 RTP/AVP 0 3 8
    	 a=rtpmap:0 PCMU/8000
    	 a=rtpmap:3 GSM/8000
    	 a=rtpmap:8 PCMA/8000
    	 a=silenceSupp:off - - - -
    	 p_c
    	 
    	  to 62.104.23.42:5060
    	  Retransmitting #2 (no NAT):
    	  INVITE sip:040123456@freenet.de SIP/2.0
    	  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	  From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	  To: <sip:040123456@freenet.de>
    	  Contact: <sip:username@192.168.1.3>
    	  Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	  CSeq: 102 INVITE
    	  User-Agent: Asterisk PBX
    	  Date: Sat, 11 Jun 2005 19:45:26 GMT
    	  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	  Content-Type: application/sdp
    	  Content-Length: 203
    	  
    	  v=0
    	  o=root 4282 4282 IN IP4 192.168.1.3
    	  s=session
    	  c=IN IP4 192.168.1.3
    	  t=0 0
    	  m=audio 11340 RTP/AVP 0 3 8
    	  a=rtpmap:0 PCMU/8000
    	  a=rtpmap:3 GSM/8000
    	  a=rtpmap:8 PCMA/8000
    	  a=silenceSupp:off - - - -
    	  p_c�
    	   to 62.104.23.42:5060
    	   Retransmitting #3 (no NAT):
    	   INVITE sip:040123456@freenet.de SIP/2.0
    	   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	   From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	   To: <sip:040123456@freenet.de>
    	   Contact: <sip:username@192.168.1.3>
    	   Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	   CSeq: 102 INVITE
    	   User-Agent: Asterisk PBX
    	   Date: Sat, 11 Jun 2005 19:45:26 GMT
    	   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	   Content-Type: application/sdp
    	   Content-Length: 203
    	   
    	   v=0
    	   o=root 4282 4282 IN IP4 192.168.1.3
    	   s=session
    	   c=IN IP4 192.168.1.3
    	   t=0 0
    	   m=audio 11340 RTP/AVP 0 3 8
    	   a=rtpmap:0 PCMU/8000
    	   a=rtpmap:3 GSM/8000
    	   a=rtpmap:8 PCMA/8000
    	   a=silenceSupp:off - - - -
    	   p_c�
    	    to 62.104.23.42:5060
    	    Retransmitting #4 (no NAT):
    	    INVITE sip:040123456@freenet.de SIP/2.0
    	    Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	    From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	    To: <sip:040123456@freenet.de>
    	    Contact: <sip:username@192.168.1.3>
    	    Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	    CSeq: 102 INVITE
    	    User-Agent: Asterisk PBX
    	    Date: Sat, 11 Jun 2005 19:45:26 GMT
    	    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	    Content-Type: application/sdp
    	    Content-Length: 203
    	    
    	    v=0
    	    o=root 4282 4282 IN IP4 192.168.1.3
    	    s=session
    	    c=IN IP4 192.168.1.3
    	    t=0 0
    	    m=audio 11340 RTP/AVP 0 3 8
    	    a=rtpmap:0 PCMU/8000
    	    a=rtpmap:3 GSM/8000
    	    a=rtpmap:8 PCMA/8000
    	    a=silenceSupp:off - - - -
    	    p_c�
    	     to 62.104.23.42:5060
    	     Retransmitting #5 (no NAT):
    	     INVITE sip:040123456@freenet.de SIP/2.0
    	     Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	     From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	     To: <sip:040123456@freenet.de>
    	     Contact: <sip:username@192.168.1.3>
    	     Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	     CSeq: 102 INVITE
    	     User-Agent: Asterisk PBX
    	     Date: Sat, 11 Jun 2005 19:45:26 GMT
    	     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    	     Content-Type: application/sdp
    	     Content-Length: 203
    	     
    	     v=0
    	     o=root 4282 4282 IN IP4 192.168.1.3
    	     s=session
    	     c=IN IP4 192.168.1.3
    	     t=0 0
    	     m=audio 11340 RTP/AVP 0 3 8
    	     a=rtpmap:0 PCMU/8000
    	     a=rtpmap:3 GSM/8000
    	     a=rtpmap:8 PCMA/8000
    	     a=silenceSupp:off - - - -
    	     p_c�
    	      to 62.104.23.42:5060
    	      Jun 11 21:45:32 WARNING[1110145968]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3 for seqno 102 (Critical Request)
    	      Reliably Transmitting:
    	      CANCEL sip:040123456@freenet.de SIP/2.0
    	      Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    	      From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    	      To: <sip:040123456@freenet.de>
    	      Contact: <sip:username@192.168.1.3>
    	      Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    	      CSeq: 102 CANCEL
    	      User-Agent: Asterisk PBX
    	      Content-Length: 0
    	      
    	       (no NAT) to 62.104.23.42:5060
    	       Scheduling destruction of call '0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3' in 15000 ms
    	         == No one is available to answer at this time
    		     -- Executing Hangup("SIP/12-7374", "") in new stack
    		       == Spawn extension (default, 9040123456, 2) exited non-zero on 'SIP/12-7374'
    		       Retransmitting #1 (no NAT):
    		       CANCEL sip:040123456@freenet.de SIP/2.0
    		       Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    		       From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    		       To: <sip:040123456@freenet.de>
    		       Contact: <sip:username@192.168.1.3>
    		       Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    		       CSeq: 102 CANCEL
    		       User-Agent: Asterisk PBX
    		       Content-Length: 0
    		       
    		       45:06Q
    		        to 62.104.23.42:5060
    			Retransmitting #2 (no NAT):
    			CANCEL sip:040123456@freenet.de SIP/2.0
    			Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    			From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    			To: <sip:040123456@freenet.de>
    			Contact: <sip:username@192.168.1.3>
    			Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    			CSeq: 102 CANCEL
    			User-Agent: Asterisk PBX
    			Content-Length: 0
    			
    			45:06Q
    			 to 62.104.23.42:5060
    			 Retransmitting #3 (no NAT):
    			 CANCEL sip:040123456@freenet.de SIP/2.0
    			 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    			 From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    			 To: <sip:040123456@freenet.de>
    			 Contact: <sip:username@192.168.1.3>
    			 Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    			 CSeq: 102 CANCEL
    			 User-Agent: Asterisk PBX
    			 Content-Length: 0
    			 
    			 45:06Q
    			  to 62.104.23.42:5060
    			  Retransmitting #4 (no NAT):
    			  CANCEL sip:040123456@freenet.de SIP/2.0
    			  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    			  From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    			  To: <sip:040123456@freenet.de>
    			  Contact: <sip:username@192.168.1.3>
    			  Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    			  CSeq: 102 CANCEL
    			  User-Agent: Asterisk PBX
    			  Content-Length: 0
    			  
    			  45:06Q
    			   to 62.104.23.42:5060
    			   Retransmitting #5 (no NAT):
    			   CANCEL sip:040123456@freenet.de SIP/2.0
    			   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    			   From: "12" <sip:username@freenet.de>;tag=as3c2deb15
    			   To: <sip:040123456@freenet.de>
    			   Contact: <sip:username@192.168.1.3>
    			   Call-ID: 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3
    			   CSeq: 102 CANCEL
    			   User-Agent: Asterisk PBX
    			   Content-Length: 0
    			   
    			   45:06Q
    			    to 62.104.23.42:5060
    			    Jun 11 21:45:38 WARNING[1110145968]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3 for seqno 102 (Non-critical Request)
    			    Destroying call '0eddd8c3661b2a7c0d709c046ed33b9b@192.168.1.3'
    			    
    
    Wo ist Fehler?
    Danke im voraus
    mfg
    Groper