Keine ausgehende Anrufe von Freenet

Groper · 13 Jun 2005

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 [email protected] 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 [email protected] 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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3431b5a4
    From: "12" <sip:[email protected]>;tag=as06a34a3d
    To: <sip:[email protected]>
    Contact: <sip:[email protected]>
    Call-ID: [email protected]
    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:[email protected] SIP/2.0
	 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3431b5a4
	 From: "12" <sip:[email protected]>;tag=as06a34a3d
	 To: <sip:[email protected]>;tag=b11cb9bb270104b49a99a995b8c68544.23ed
	 Contact: <sip:[email protected]>
	 Call-ID: [email protected]
	 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:[email protected] SIP/2.0
	  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK6a8bcaf0
	  From: "12" <sip:[email protected]>;tag=as06a34a3d
	  To: <sip:[email protected]>
	  Contact: <sip:[email protected]>
	  Call-ID: [email protected]
	  CSeq: 103 INVITE
	  User-Agent: Asterisk PBX
	  Proxy-Authorization: Digest username="030123456", realm="sipgate.de", algorithm=MD5, uri="sip:[email protected]", 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:[email protected];ftag=as06a34a3d;lr=on>
	   list_route: hop: <sip:[email protected];ftag=as06a34a3d;lr=on>
	   list_route: hop: <sip:[email protected]>
	   set_destination: Parsing <sip:[email protected];ftag=as06a34a3d;lr=on> for address/port to send to
	   set_destination: set destination to 217.10.79.9, port 5060
	   Transmitting:
	   ACK sip:[email protected] SIP/2.0
	   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK00584e38
	   Route: <sip:[email protected];ftag=as06a34a3d;lr=on>,<sip:[email protected]>
	   From: "12" <sip:[email protected]>;tag=as06a34a3d
	   To: <sip:[email protected]>;tag=as7632f45f
	   Contact: <sip:[email protected]>
	   Call-ID: [email protected]
	   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:[email protected];ftag=as7632f45f;lr=on>
		    Record-Route: <sip:[email protected];ftag=as7632f45f;lr=on>
		    From: <sip:[email protected]>;tag=as7632f45f
		    To: "12" <sip:[email protected]>;tag=as06a34a3d
		    Call-ID: [email protected]
		    CSeq: 102 BYE
		    User-Agent: Asterisk PBX
		    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
		    Contact: <sip:[email protected]>
		    Content-Length: 0
		    
		    
		     to 217.10.79.9:5060
		       == Spawn extension (default, 1040123456, 1) exited non-zero on 'SIP/12-66b9'
		       Destroying call '[email protected]'

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:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
    From: "12" <sip:[email protected]>;tag=as3c2deb15
    To: <sip:[email protected]>
    Contact: <sip:[email protected]>
    Call-ID: [email protected]
    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:[email protected] SIP/2.0
	 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	 From: "12" <sip:[email protected]>;tag=as3c2deb15
	 To: <sip:[email protected]>
	 Contact: <sip:[email protected]>
	 Call-ID: [email protected]
	 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:[email protected] SIP/2.0
	  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	  From: "12" <sip:[email protected]>;tag=as3c2deb15
	  To: <sip:[email protected]>
	  Contact: <sip:[email protected]>
	  Call-ID: [email protected]
	  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:[email protected] SIP/2.0
	   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	   From: "12" <sip:[email protected]>;tag=as3c2deb15
	   To: <sip:[email protected]>
	   Contact: <sip:[email protected]>
	   Call-ID: [email protected]
	   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:[email protected] SIP/2.0
	    Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	    From: "12" <sip:[email protected]>;tag=as3c2deb15
	    To: <sip:[email protected]>
	    Contact: <sip:[email protected]>
	    Call-ID: [email protected]
	    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:[email protected] SIP/2.0
	     Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	     From: "12" <sip:[email protected]>;tag=as3c2deb15
	     To: <sip:[email protected]>
	     Contact: <sip:[email protected]>
	     Call-ID: [email protected]
	     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 [email protected] for seqno 102 (Critical Request)
	      Reliably Transmitting:
	      CANCEL sip:[email protected] SIP/2.0
	      Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
	      From: "12" <sip:[email protected]>;tag=as3c2deb15
	      To: <sip:[email protected]>
	      Contact: <sip:[email protected]>
	      Call-ID: [email protected]
	      CSeq: 102 CANCEL
	      User-Agent: Asterisk PBX
	      Content-Length: 0
	      
	       (no NAT) to 62.104.23.42:5060
	       Scheduling destruction of call '[email protected]' 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:[email protected] SIP/2.0
		       Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
		       From: "12" <sip:[email protected]>;tag=as3c2deb15
		       To: <sip:[email protected]>
		       Contact: <sip:[email protected]>
		       Call-ID: [email protected]
		       CSeq: 102 CANCEL
		       User-Agent: Asterisk PBX
		       Content-Length: 0
		       
		       45:06Q
		        to 62.104.23.42:5060
			Retransmitting #2 (no NAT):
			CANCEL sip:[email protected] SIP/2.0
			Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
			From: "12" <sip:[email protected]>;tag=as3c2deb15
			To: <sip:[email protected]>
			Contact: <sip:[email protected]>
			Call-ID: [email protected]
			CSeq: 102 CANCEL
			User-Agent: Asterisk PBX
			Content-Length: 0
			
			45:06Q
			 to 62.104.23.42:5060
			 Retransmitting #3 (no NAT):
			 CANCEL sip:[email protected] SIP/2.0
			 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
			 From: "12" <sip:[email protected]>;tag=as3c2deb15
			 To: <sip:[email protected]>
			 Contact: <sip:[email protected]>
			 Call-ID: [email protected]
			 CSeq: 102 CANCEL
			 User-Agent: Asterisk PBX
			 Content-Length: 0
			 
			 45:06Q
			  to 62.104.23.42:5060
			  Retransmitting #4 (no NAT):
			  CANCEL sip:[email protected] SIP/2.0
			  Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
			  From: "12" <sip:[email protected]>;tag=as3c2deb15
			  To: <sip:[email protected]>
			  Contact: <sip:[email protected]>
			  Call-ID: [email protected]
			  CSeq: 102 CANCEL
			  User-Agent: Asterisk PBX
			  Content-Length: 0
			  
			  45:06Q
			   to 62.104.23.42:5060
			   Retransmitting #5 (no NAT):
			   CANCEL sip:[email protected] SIP/2.0
			   Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK59283e60
			   From: "12" <sip:[email protected]>;tag=as3c2deb15
			   To: <sip:[email protected]>
			   Contact: <sip:[email protected]>
			   Call-ID: [email protected]
			   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 [email protected] for seqno 102 (Non-critical Request)
			    Destroying call '[email protected]'

Wo ist Fehler?
Danke im voraus
mfg
Groper

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