Script iniziato su mer 10 ago 2005 17:14:30 CEST ]0;root@pbx-test:~[root@pbx-test ~]# exitasterisk -r Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD currently running on pbx-test (pid = 4392) pbx-test*CLI> Verbosity is at least 3 pbx-test*CLI> sip debug pbx-test*CLI> SIP Debugging enabled pbx-test*CLI> <-- SIP read from 192.168.2.101:5060: INVITE sip:0458005593@192.168.2.121 SIP/2.0v: SIP/2.0/UDP 192.168.2.101;rport;branch=z9hG4bKc0a802650000044542fa19e80000366c00001350l: 294m: i: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101c: application/sdpCSeq: 1 INVITEf: "Alessandro";tag=2936490429210Max-Forwards: 70t: User-Agent: SJphone/1.60.289a (SJ Labs)v=0o=- 3332675688 3332675688 IN IP4 192.168.2.101s=SJphonec=IN IP4 192.168.2.101t=0 0a=direction:activem=audio 49208 RTP/AVP 97 3 8 0 101a=rtpmap:97 iLBC/8000a=rtpmap:3 GSM/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,16 --- (11 headers 13 lines)--- Using INVITE request as basis request - 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101 Sending to 192.168.2.101 : 5060 (non-NAT) Reliably Transmitting (NAT) to 192.168.2.101:5060: SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bKc0a802650000044542fa19e80000366c00001350;received=192.168.2.101;rport=5060From: "Alessandro";tag=2936490429210To: ;tag=as3b9c553cCall-ID: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Proxy-Authenticate: Digest realm="kleis.it", nonce="05cc89db" Content-Length: 0 --- Scheduling destruction of call '9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101' in 15000 ms Found user '304' pbx-test*CLI> <-- SIP read from 192.168.2.101:5060: ACK sip:0458005593@192.168.2.121 SIP/2.0v: SIP/2.0/UDP 192.168.2.101;rport;branch=z9hG4bKc0a802650000044542fa19e80000366c00001350l: 0i: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101CSeq: 1 ACKf: "Alessandro";tag=2936490429210Max-Forwards: 70t: ;tag=as3b9c553cUser-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- <-- SIP read from 192.168.2.101:5060: INVITE sip:0458005593@192.168.2.121 SIP/2.0v: SIP/2.0/UDP 192.168.2.101;rport;branch=z9hG4bKc0a802650000044542fa19e80000576200001351l: 294m: i: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101c: application/sdpCSeq: 2 INVITEf: "Alessandro";tag=2936490429210Max-Forwards: 70t: User-Agent: SJphone/1.60.289a (SJ Labs)Proxy-Authorization: Digest username="304",realm="kleis.it",nonce="05cc89db",uri="sip:0458005593@192.168.2.121",response="d9833c6a4db27a6ad1e4e910f1877310"v=0o=- 3332675688 3332675688 IN IP4 192.168.2.101s=SJphonec=IN IP4 192.168.2.101t=0 0a=direction:activem=audio 49208 RTP/AVP 97 3 8 0 101a=rtpmap:97 iLBC/8000a=rtpmap:3 GSM/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,16 --- (12 headers 13 lines)--- Using INVITE request as basis request - 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101 Sending to 192.168.2.101 : 5060 (NAT) Found user '304' Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.101:49208 Found description format iLBC Found description format GSM Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/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 0458005593 in sip-kleis-in pbx-test*CLI> Reliably Transmitting (NAT) to 192.168.2.101:5060: SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bKc0a802650000044542fa19e80000576200001351;received=192.168.2.101;rport=5060From: "Alessandro";tag=2936490429210To: ;tag=as3b9c553cCall-ID: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101CSeq: 2 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Content-Length: 0 --- pbx-test*CLI> <-- SIP read from 192.168.2.101:5060: ACK sip:0458005593@192.168.2.121 SIP/2.0v: SIP/2.0/UDP 192.168.2.101;rport;branch=z9hG4bKc0a802650000044542fa19e80000576200001351l: 0i: 9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101CSeq: 2 ACKf: "Alessandro";tag=2936490429210Max-Forwards: 70t: ;tag=as3b9c553cUser-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- Destroying call '9EBCE5CE-437D-4CE3-BC71-4E81078D7824@192.168.2.101' pbx-test*CLI> <-- SIP read from 192.168.2.105:5060: OPTIONS sip:192.168.2.121 SIP/2.0Content-Length: 0Call-ID: BFB0C738-616D-4C89-9139-B168BFC12BCF@192.168.2.105From: ;tag=298516716815CSeq: 1080 OPTIONSMax-Forwards: 70To: Via: SIP/2.0/UDP 192.168.2.105;rport;branch=z9hG4bKc0a802690131c9b142fa19e900002fc8000011cb --- (8 headers 0 lines)--- Looking for 192.168.2.121 in default Transmitting (no NAT) to 192.168.2.105:5060: SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.2.105;branch=z9hG4bKc0a802690131c9b142fa19e900002fc8000011cbFrom: ;tag=298516716815To: ;tag=as1ede908aCall-ID: BFB0C738-616D-4C89-9139-B168BFC12BCF@192.168.2.105CSeq: 1080 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Accept: application/sdpContent-Length: 0 --- Destroying call 'BFB0C738-616D-4C89-9139-B168BFC12BCF@192.168.2.105' pbx-test*CLI> <-- SIP read from 192.168.2.101:5060: OPTIONS sip:192.168.2.121 SIP/2.0v: SIP/2.0/UDP 192.168.2.101;rport;branch=z9hG4bKc0a802650000001042fa19eb0000367600001353l: 0i: 379C79A2-E6FB-42EC-BBE6-2653A9B66015@192.168.2.101CSeq: 933 OPTIONSf: ;tag=2936823919886Max-Forwards: 70t: --- (8 headers 0 lines)--- Looking for 192.168.2.121 in default Transmitting (no NAT) to 192.168.2.101:5060: SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bKc0a802650000001042fa19eb0000367600001353From: ;tag=2936823919886To: ;tag=as58f9bc8dCall-ID: 379C79A2-E6FB-42EC-BBE6-2653A9B66015@192.168.2.101CSeq: 933 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Accept: application/sdpContent-Length: 0 --- Destroying call '379C79A2-E6FB-42EC-BBE6-2653A9B66015@192.168.2.101' pbx-test*CLI> exit]0;root@pbx-test:~[root@pbx-test ~]# exit Script effettuato su mer 10 ago 2005 17:14:59 CEST