Call Declined 603?

[Maui]

Hallo zusammen,

ich habe für einen Bekannten einen kleinen Asterisk Server mit FreePBX aufgesetzt und versuche nun diesesn zu erreichen. Mein Telefon sagt mir aber direkt nach dem wählen "Teilnehmer besetzt" und das obwohl die Sipgate Nr. auf einer Testwarteschleife terminieren soll.

Hier mal ein Ausschnit des Sip Logs auf dem Asterisk Server meines bekannten:

<--- SIP read from 217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as048e5e41>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as048e5e41>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKc583.95842415.0
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKc583.95842415.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK63f334e2
Via: SIP/2.0/UDP 85.179.200.186:5060;branch=z9hG4bK63f334e2;rport=5060
From: "Sebastian Mauer" <sip:[email protected]>;tag=as048e5e41
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 31503 31504 IN IP4 85.179.200.186
s=session
c=IN IP4 85.179.200.186
t=0 0
m=audio 16568 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 12 lines) ---
Sending to 217.10.79.9 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'sipgate'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 85.179.200.186:16568
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 85.179.200.186:16568
Looking for 5048325 in from-sip-external (domain 78.46.85.51)
list_route: hop: <sip:217.10.79.9;lr=on;ftag=as048e5e41>
list_route: hop: <sip:172.20.40.1;lr=on>
list_route: hop: <sip:217.10.79.9;lr=on;ftag=as048e5e41>

<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKc583.95842415.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKc583.95842415.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK63f334e2
Via: SIP/2.0/UDP 85.179.200.186:5060;branch=z9hG4bK63f334e2;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as048e5e41>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as048e5e41>
From: "Sebastian Mauer" <sip:[email protected]>;tag=as048e5e41
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
wilson*CLI>
<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKc583.95842415.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKc583.95842415.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK63f334e2
Via: SIP/2.0/UDP 85.179.200.186:5060;branch=z9hG4bK63f334e2;rport=5060
From: "Sebastian Mauer" <sip:[email protected]>;tag=as048e5e41
To: <sip:[email protected]>;tag=as2162b60b
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

Was läuft da nur schief. Der Asterisk Server hängt ohne NAT direkt an einer festen IP

 

[himpierre]

In welchem Kontext landet der sipgate trunk? In welchem befindet sich Deine Testschleife?