Apr 7 11:16:18 VERBOSE[10085]: Asterisk Event Logger restarted Apr 7 11:16:23 VERBOSE[6513]: Sip read: REGISTER sip:62.207.132.15:5061 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKcc38.20044c07.0 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;rport=5060;branch=z9hG4bK_0.5538774f From: ;tag=1112865390.328 To: Call-ID: 79c8ee68 CSeq: 14256 REGISTER Contact: ;expires=1800;action=proxy;methods="INVITE,BYE,ACK,CANCEL,SUBSCRIBE,MESSAGE,NOTIFY" Max-Forwards: 5 Content-length: 0 Apr 7 11:16:23 VERBOSE[6513]: 11 headers, 0 lines Apr 7 11:16:23 DEBUG[6513]: Allocating new SIP call for 79c8ee68 Apr 7 11:16:23 VERBOSE[6513]: Using latest request as basis request Apr 7 11:16:23 VERBOSE[6513]: Sending to 62.207.132.15 : 5060 (NAT) Apr 7 11:16:23 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKcc38.20044c07.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.5538774f From: ;tag=1112865390.328 To: ;tag=as49ebbe53 Call-ID: 79c8ee68 CSeq: 14256 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:23 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKcc38.20044c07.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.5538774f From: ;tag=1112865390.328 To: ;tag=as49ebbe53 Call-ID: 79c8ee68 CSeq: 14256 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="sip.cheapnet.it", nonce="28a41409" Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:23 VERBOSE[6513]: Scheduling destruction of call '79c8ee68' in 15000 ms Apr 7 11:16:23 VERBOSE[6513]: Sip read: REGISTER sip:62.207.132.15:5061 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKdc38.fa7eafe1.0 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;rport=5060;branch=z9hG4bK_0.5ef80beb From: ;tag=1112865390.390 To: Call-ID: 79c8ee68 CSeq: 14257 REGISTER Contact: ;expires=1800;action=proxy;methods="INVITE,BYE,ACK,CANCEL,SUBSCRIBE,MESSAGE,NOTIFY" Max-Forwards: 5 Authorization: Digest username="6554000000",realm="sip.cheapnet.it",nonce="28a41409",uri="sip:sip.cheapnet.it",response="55d697243a4ebed89ebcee19ef3d9483" Content-length: 0 Apr 7 11:16:23 VERBOSE[6513]: 12 headers, 0 lines Apr 7 11:16:23 VERBOSE[6513]: Using latest request as basis request Apr 7 11:16:23 VERBOSE[6513]: Sending to 62.207.132.15 : 5060 (NAT) Apr 7 11:16:23 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKdc38.fa7eafe1.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.5ef80beb From: ;tag=1112865390.390 To: ;tag=as49ebbe53 Call-ID: 79c8ee68 CSeq: 14257 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:23 DEBUG[6513]: Allocating new SIP call for (null) Apr 7 11:16:23 VERBOSE[6513]: 11 headers, 0 lines Apr 7 11:16:23 VERBOSE[6513]: Reliably Transmitting: OPTIONS sip:192.168.2.75:5060 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK173c9a01 From: "asterisk" ;tag=as1886a565 To: Contact: Call-ID: 04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 07 Apr 2005 09:16:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 62.207.132.15:5060 Apr 7 11:16:23 VERBOSE[6513]: -- Registered SIP '6554000000' at 62.207.132.15 port 5060 expires 1800 Apr 7 11:16:23 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKdc38.fa7eafe1.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.5ef80beb Record-Route: From: ;tag=1112865390.390 To: ;tag=as49ebbe53 Call-ID: 79c8ee68 CSeq: 14257 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 1800 Contact: ;expires=1800 Date: Thu, 07 Apr 2005 09:16:23 GMT Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:23 VERBOSE[6513]: Scheduling destruction of call '79c8ee68' in 15000 ms Apr 7 11:16:23 VERBOSE[6513]: Sip read: OPTIONS sip:62.207.132.15:5061 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bK47fa.e0beadd2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK173c9a01 From: "asterisk" ;tag=as1886a565 To: Contact: Call-ID: 04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 07 Apr 2005 09:16:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 Apr 7 11:16:23 VERBOSE[6513]: 13 headers, 0 lines Apr 7 11:16:23 VERBOSE[6513]: Looking for 62.207.132.15:5061 in messagenet-in Apr 7 11:16:23 VERBOSE[6513]: Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bK47fa.e0beadd2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK173c9a01 From: "asterisk" ;tag=as1886a565 To: ;tag=as1886a565 Call-ID: 04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Accept: application/sdp Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:23 VERBOSE[6513]: Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK173c9a01 From: "asterisk" ;tag=as1886a565 To: ;tag=as1886a565 Call-ID: 04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Accept: application/sdp Content-Length: 0 Apr 7 11:16:23 VERBOSE[6513]: 11 headers, 0 lines Apr 7 11:16:23 DEBUG[6513]: Stopping retransmission on '04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15' of Request 102: Found Apr 7 11:16:23 VERBOSE[6513]: Destroying call '04e9aabb75db687b2c6a9b2e52fb5e85@62.207.132.15' Apr 7 11:16:25 VERBOSE[6513]: Sip read: INVITE sip:0585255362@62.207.132.15:5061;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKb7b4.109c7703.0 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;rport=5060;branch=z9hG4bK_0.a61138e8 From: ;tag=1112865392.734 To: Call-ID: 90977af5 CSeq: 14258 INVITE Max-Forwards: 5 Contact: Session-Expires: 30 Content-length: 135 Content-type: application/sdp v=0 o=username 18638 770 IN IP4 192.168.2.75 s=Sip Call c=IN IP4 192.168.2.75 t=0 0 m=audio 5004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Apr 7 11:16:25 VERBOSE[6513]: 13 headers, 7 lines Apr 7 11:16:25 DEBUG[6513]: Allocating new SIP call for 90977af5 Apr 7 11:16:25 VERBOSE[6513]: Using latest request as basis request Apr 7 11:16:25 VERBOSE[6513]: Sending to 62.207.132.15 : 5060 (NAT) Apr 7 11:16:25 DEBUG[6513]: Setting NAT on RTP to 4 Apr 7 11:16:25 VERBOSE[6513]: Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKb7b4.109c7703.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a61138e8 From: ;tag=1112865392.734 To: ;tag=as5a2023a7 Call-ID: 90977af5 CSeq: 14258 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="sip.cheapnet.it", nonce="4338d524" Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:25 VERBOSE[6513]: Scheduling destruction of call '90977af5' in 15000 ms Apr 7 11:16:25 VERBOSE[6513]: Found user '6554000000' Apr 7 11:16:25 VERBOSE[6513]: Sip read: ACK sip:0585255362@62.207.132.15:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKb7b4.109c7703.0 From: ;tag=1112865392.734 Call-ID: 90977af5 To: ;tag=as5a2023a7 CSeq: 14258 ACK User-Agent: Sip EXpress router(0.9.0 (i386/linux)) Content-Length: 0 Apr 7 11:16:25 VERBOSE[6513]: 8 headers, 0 lines Apr 7 11:16:25 DEBUG[6513]: Stopping retransmission on '90977af5' of Response 14258: Found Apr 7 11:16:25 VERBOSE[6513]: Sip read: INVITE sip:0585255362@62.207.132.15:5061;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;rport=5060;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: Call-ID: 90977af5 CSeq: 14259 INVITE Max-Forwards: 5 Contact: Session-Expires: 30 Proxy-Authorization: Digest username="6554000000",realm="sip.cheapnet.it",nonce="4338d524",uri="sip:0585255362@sip.cheapnet.it;user=phone",response="328777d49db4a09bd8eb9e83d5f72593" Content-length: 135 Content-type: application/sdp v=0 o=username 18638 771 IN IP4 192.168.2.75 s=Sip Call c=IN IP4 192.168.2.75 t=0 0 m=audio 5004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Apr 7 11:16:25 VERBOSE[6513]: 14 headers, 7 lines Apr 7 11:16:25 VERBOSE[6513]: Using latest request as basis request Apr 7 11:16:25 VERBOSE[6513]: Sending to 62.207.132.15 : 5060 (NAT) Apr 7 11:16:25 DEBUG[6513]: Setting NAT on RTP to 4 Apr 7 11:16:25 VERBOSE[6513]: Found user '6554000000' Apr 7 11:16:25 VERBOSE[6513]: Found RTP audio format 0 Apr 7 11:16:25 VERBOSE[6513]: Peer audio RTP is at port 192.168.2.75:5004 Apr 7 11:16:25 DEBUG[6513]: Peer audio RTP is at port 192.168.2.75:5004 Apr 7 11:16:25 VERBOSE[6513]: Found description format PCMU Apr 7 11:16:25 VERBOSE[6513]: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Apr 7 11:16:25 VERBOSE[6513]: Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Apr 7 11:16:25 DEBUG[6513]: Check for res for 6554000000 Apr 7 11:16:25 DEBUG[6513]: Call from user '6554000000' is 1 out of 0 Apr 7 11:16:25 VERBOSE[6513]: Looking for 0585255362 in cheapdemo Apr 7 11:16:25 DEBUG[6513]: build_route: Record-Route hop: Apr 7 11:16:25 DEBUG[6513]: build_route: Contact hop: Apr 7 11:16:25 VERBOSE[6513]: list_route: hop: Apr 7 11:16:25 VERBOSE[6513]: list_route: hop: Apr 7 11:16:25 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: ;tag=as335ade2f Call-ID: 90977af5 CSeq: 14259 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:25 DEBUG[10092]: Launching 'SetVar' Apr 7 11:16:25 VERBOSE[10092]: -- Executing SetVar("SIP/6554000000-b52d", "LIMIT_WARNING_FILE=beep") in new stack Apr 7 11:16:25 DEBUG[10092]: Launching 'SetCallerID' Apr 7 11:16:25 VERBOSE[10092]: -- Executing SetCallerID("SIP/6554000000-b52d", "5301392") in new stack Apr 7 11:16:25 DEBUG[10092]: Launching 'Ringing' Apr 7 11:16:25 VERBOSE[10092]: -- Executing Ringing("SIP/6554000000-b52d", "") in new stack Apr 7 11:16:25 VERBOSE[10092]: Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: ;tag=as335ade2f Call-ID: 90977af5 CSeq: 14259 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:25 DEBUG[10092]: Driver for channel 'SIP/6554000000-b52d' does not support indication 3, emulating it Apr 7 11:16:25 DEBUG[10092]: Prodding channel 'SIP/6554000000-b52d' Apr 7 11:16:25 VERBOSE[10092]: We're at 62.207.132.15 port 18652 Apr 7 11:16:25 VERBOSE[10092]: Answering with preferred capability 0x2 (gsm) Apr 7 11:16:25 VERBOSE[10092]: Answering with preferred capability 0x8 (alaw) Apr 7 11:16:25 VERBOSE[10092]: Answering with preferred capability 0x4 (ulaw) Apr 7 11:16:25 VERBOSE[10092]: Transmitting (NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: ;tag=as335ade2f Call-ID: 90977af5 CSeq: 14259 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 10092 10092 IN IP4 62.207.132.15 s=session c=IN IP4 62.207.132.15 t=0 0 m=audio 18652 RTP/AVP 3 8 0 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - to 62.207.132.15:5060 Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/6554000000-b52d to write format slin Apr 7 11:16:25 DEBUG[10092]: Launching 'Dial' Apr 7 11:16:25 VERBOSE[10092]: -- Executing Dial("SIP/6554000000-b52d", "SIP/0585255362@messagenet||L(600000:10000:7000)") in new stack Apr 7 11:16:25 DEBUG[10092]: SIMPLE DIAL (NO URL) Apr 7 11:16:25 VERBOSE[10092]: -- Limit Data: Apr 7 11:16:25 VERBOSE[10092]: -- timelimit=600000 Apr 7 11:16:25 VERBOSE[10092]: -- play_warning=10000 Apr 7 11:16:25 VERBOSE[10092]: -- play_to_caller=yes Apr 7 11:16:25 VERBOSE[10092]: -- play_to_callee=no Apr 7 11:16:25 VERBOSE[10092]: -- warning_freq=7000 Apr 7 11:16:25 VERBOSE[10092]: -- start_sound=UNDEF Apr 7 11:16:25 VERBOSE[10092]: -- warning_sound=beep Apr 7 11:16:25 VERBOSE[10092]: -- end_sound=UNDEF Apr 7 11:16:25 DEBUG[10092]: Allocating new SIP call for (null) Apr 7 11:16:25 DEBUG[10092]: Setting NAT on RTP to 0 Apr 7 11:16:25 DEBUG[10092]: Outgoing Call for 0585255362 Apr 7 11:16:25 DEBUG[10092]: 0585255362 is not a local user Apr 7 11:16:25 VERBOSE[10092]: We're at 62.207.132.15 port 14748 Apr 7 11:16:25 VERBOSE[10092]: Answering with capability 0x8 (alaw) Apr 7 11:16:25 VERBOSE[10092]: Answering with non-codec capability 0x1 (telephone-event) Apr 7 11:16:25 VERBOSE[10092]: 12 headers, 10 lines Apr 7 11:16:25 VERBOSE[10092]: Reliably Transmitting: INVITE sip:0585255362@sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK3ef15ada From: "5301392" ;tag=as031911c7 To: Contact: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 07 Apr 2005 09:16:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 10092 10092 IN IP4 62.207.132.15 s=session c=IN IP4 62.207.132.15 t=0 0 m=audio 14748 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 212.97.59.76:5061 Apr 7 11:16:25 VERBOSE[10092]: -- Called 0585255362@messagenet Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/6554000000-b52d to write format ulaw Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/messagenet-381e to read format ulaw Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/6554000000-b52d to write format ulaw Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/messagenet-381e to write format alaw Apr 7 11:16:25 DEBUG[10092]: Set channel SIP/6554000000-b52d to read format alaw Apr 7 11:16:25 VERBOSE[6513]: Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK3ef15ada From: "5301392" ;tag=as031911c7 To: ;tag=7bf1846f810bb99b120843b93d982c7d.8816 Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sip.messagenet.it", nonce="4254fb95afcdfa5cc884517ad9ccf0bdbc1959e9" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=5555 req_src_ip=62.207.132.15 req_src_port=5061 in_uri=sip:0585255362@sip.messagenet.it:5061 out_uri=sip:0585255362@sip.messagenet.it:5061 via_cnt==1" Apr 7 11:16:25 VERBOSE[6513]: 10 headers, 0 lines Apr 7 11:16:25 DEBUG[6513]: Acked pending invite 102 Apr 7 11:16:25 DEBUG[6513]: Stopping retransmission on '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' of Request 102: Found Apr 7 11:16:25 VERBOSE[6513]: Transmitting: ACK sip:0585255362@sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK3ef15ada From: "5301392" ;tag=as031911c7 To: ;tag=7bf1846f810bb99b120843b93d982c7d.8816 Contact: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 212.97.59.76:5061 Apr 7 11:16:25 VERBOSE[6513]: We're at 62.207.132.15 port 14748 Apr 7 11:16:25 VERBOSE[6513]: Answering with capability 0x8 (alaw) Apr 7 11:16:25 VERBOSE[6513]: Answering with non-codec capability 0x1 (telephone-event) Apr 7 11:16:25 VERBOSE[6513]: Reliably Transmitting: INVITE sip:0585255362@sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: Contact: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="5301392", realm="sip.messagenet.it", algorithm=MD5, uri="sip:0585255362@sip.messagenet.it:5061", nonce="4254fb95afcdfa5cc884517ad9ccf0bdbc1959e9", response="37c2d76f9d8dc060355f88842074fdb8", opaque="" Date: Thu, 07 Apr 2005 09:16:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 10092 10093 IN IP4 62.207.132.15 s=session c=IN IP4 62.207.132.15 t=0 0 m=audio 14748 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 212.97.59.76:5061 Apr 7 11:16:25 VERBOSE[6513]: Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=5531 req_src_ip=62.207.132.15 req_src_port=5061 in_uri=sip:0585255362@sip.messagenet.it:5061 out_uri=sip:0585255362@212.97.59.75:5060 via_cnt==1" Apr 7 11:16:25 VERBOSE[6513]: 9 headers, 0 lines Apr 7 11:16:25 DEBUG[6513]: (Provisional) Stopping retransmission (but retaining packet) on '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' Request 103: Found Apr 7 11:16:25 VERBOSE[6513]: Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: ;tag=as504518f6 Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 15943 15943 IN IP4 212.97.59.75 s=session c=IN IP4 212.97.59.75 t=0 0 m=audio 12750 RTP/AVP 18 97 8 101 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Apr 7 11:16:25 VERBOSE[6513]: 11 headers, 12 lines Apr 7 11:16:25 DEBUG[6513]: (Provisional) Stopping retransmission (but retaining packet) on '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' Request 103: Found Apr 7 11:16:25 VERBOSE[6513]: Found RTP audio format 18 Apr 7 11:16:25 VERBOSE[6513]: Found RTP audio format 97 Apr 7 11:16:25 VERBOSE[6513]: Found RTP audio format 8 Apr 7 11:16:25 VERBOSE[6513]: Found RTP audio format 101 Apr 7 11:16:25 VERBOSE[6513]: Peer audio RTP is at port 212.97.59.75:12750 Apr 7 11:16:25 DEBUG[6513]: Peer audio RTP is at port 212.97.59.75:12750 Apr 7 11:16:25 VERBOSE[6513]: Found description format G729 Apr 7 11:16:25 VERBOSE[6513]: Found description format iLBC Apr 7 11:16:25 VERBOSE[6513]: Found description format PCMA Apr 7 11:16:25 VERBOSE[6513]: Found description format telephone-event Apr 7 11:16:25 VERBOSE[6513]: Capabilities: us - 0x8 (alaw), peer - audio=0x508 (alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) Apr 7 11:16:25 VERBOSE[6513]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Apr 7 11:16:25 VERBOSE[10092]: -- SIP/messagenet-381e is making progress passing it to SIP/6554000000-b52d Apr 7 11:16:25 DEBUG[10092]: Ooh, format changed from unknown to ulaw Apr 7 11:16:30 VERBOSE[6513]: Sip read: CANCEL sip:0585255362@62.207.132.15:5061;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;rport=5060;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: Call-ID: 90977af5 CSeq: 14259 CANCEL Max-Forwards: 5 Apr 7 11:16:30 VERBOSE[6513]: 9 headers, 0 lines Apr 7 11:16:30 VERBOSE[6513]: Sending to 62.207.132.15 : 5060 (NAT) Apr 7 11:16:30 VERBOSE[6513]: Reliably Transmitting (NAT): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a8a21029 From: ;tag=1112865392.734 To: ;tag=as335ade2f Call-ID: 90977af5 CSeq: 14259 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:30 VERBOSE[6513]: Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0;received=62.207.132.15;rport=5060 Via: SIP/2.0/UDP 192.168.2.75:5060;received=80.67.114.32;branch=z9hG4bK_0.a8a21029 Record-Route: From: ;tag=1112865392.734 To: ;tag=as335ade2f Call-ID: 90977af5 CSeq: 14259 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 62.207.132.15:5060 Apr 7 11:16:30 DEBUG[10092]: Hanging up channel 'SIP/messagenet-381e' Apr 7 11:16:30 DEBUG[10092]: sip_hangup(SIP/messagenet-381e) Apr 7 11:16:30 DEBUG[10092]: update_user_counter(0585255362) - decrement outUse counter Apr 7 11:16:30 DEBUG[10092]: 0585255362 is not a local user Apr 7 11:16:30 VERBOSE[10092]: Reliably Transmitting: CANCEL sip:0585255362@sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: Contact: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username="5301392", realm="sip.messagenet.it", algorithm=MD5, uri="sip:0585255362@212.97.59.75", nonce="4254fb95afcdfa5cc884517ad9ccf0bdbc1959e9", response="e222c62672fcf73b99afc5145ab2319d", opaque="" Content-Length: 0 (no NAT) to 212.97.59.76:5061 Apr 7 11:16:30 VERBOSE[10092]: Scheduling destruction of call '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' in 15000 ms Apr 7 11:16:30 DEBUG[10092]: Exiting with DIALSTATUS=CANCEL. Apr 7 11:16:30 DEBUG[10092]: Spawn extension (cheapdemo,0585255362,4) exited non-zero on 'SIP/6554000000-b52d' Apr 7 11:16:30 VERBOSE[6513]: Sip read: ACK sip:0585255362@62.207.132.15:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15;branch=z9hG4bKa7b4.87d4d9a7.0 From: ;tag=1112865392.734 Call-ID: 90977af5 To: ;tag=as335ade2f CSeq: 14259 ACK User-Agent: Sip EXpress router(0.9.0 (i386/linux)) Content-Length: 0 Apr 7 11:16:30 VERBOSE[6513]: 8 headers, 0 lines Apr 7 11:16:30 DEBUG[6513]: Failed to grab lock, trying again... Apr 7 11:16:30 DEBUG[10092]: cdr_mysql: inserting a CDR record. Apr 7 11:16:30 DEBUG[10092]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2005-04-07 11:16:25','5301392','5301392','0585255362','cheapdemo', 'SIP/6554000000-b52d','SIP/messagenet-381e','Dial','SIP/0585255362@messagenet||L(600000:10000:7000)',5,0,'NO ANSWER',3,'','') Apr 7 11:16:30 DEBUG[10092]: Hanging up channel 'SIP/6554000000-b52d' Apr 7 11:16:30 DEBUG[10092]: sip_hangup(SIP/6554000000-b52d) Apr 7 11:16:30 DEBUG[10092]: update_user_counter(6554000000) - decrement inUse counter Apr 7 11:16:30 DEBUG[6513]: Stopping retransmission on '90977af5' of Response 14259: Found Apr 7 11:16:30 VERBOSE[6513]: Destroying call '90977af5' Apr 7 11:16:30 VERBOSE[6513]: Sip read: SIP/2.0 200 cancelling Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: ;tag=e5bd87eec7a3ecc921ebf3d25f7f071b-9eba Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 CANCEL Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=5537 req_src_ip=62.207.132.15 req_src_port=5061 in_uri=sip:0585255362@sip.messagenet.it:5061 out_uri=sip:0585255362@212.97.59.75:5060 via_cnt==1" Apr 7 11:16:30 VERBOSE[6513]: 9 headers, 0 lines Apr 7 11:16:30 DEBUG[6513]: Acked pending invite 103 Apr 7 11:16:30 DEBUG[6513]: Stopping retransmission on '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' of Request 103: Found Apr 7 11:16:30 VERBOSE[6513]: Sip read: SIP/2.0 487 Request cancelled Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: ;tag=e5bd87eec7a3ecc921ebf3d25f7f071b-9eba Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=5537 req_src_ip=62.207.132.15 req_src_port=5061 in_uri=sip:0585255362@sip.messagenet.it:5061 out_uri=sip:0585255362@212.97.59.75:5060 via_cnt==1" Apr 7 11:16:30 VERBOSE[6513]: 9 headers, 0 lines Apr 7 11:16:30 DEBUG[6513]: Stopping retransmission on '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' of Request 103: Found Apr 7 11:16:30 DEBUG[6513]: 0585255362 is not a local user Apr 7 11:16:30 VERBOSE[6513]: Transmitting: ACK sip:0585255362@sip.messagenet.it:5061 SIP/2.0 Via: SIP/2.0/UDP 62.207.132.15:5061;branch=z9hG4bK4f076d41 From: "5301392" ;tag=as031911c7 To: ;tag=e5bd87eec7a3ecc921ebf3d25f7f071b-9eba Contact: Call-ID: 20bf91ae7ae592b87734ae623b5929cb@62.207.132.15 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 212.97.59.76:5061 Apr 7 11:16:30 VERBOSE[6513]: Destroying call '20bf91ae7ae592b87734ae623b5929cb@62.207.132.15' Apr 7 11:16:31 DEBUG[6513]: Auto destroying call '4185345834@192.168.2.30' Apr 7 11:16:31 VERBOSE[6513]: Destroying call '4185345834@192.168.2.30' Apr 7 11:16:31 DEBUG[6513]: Auto destroying call '2416831404@192.168.2.30' Apr 7 11:16:31 VERBOSE[6513]: Destroying call '2416831404@192.168.2.30' Apr 7 11:16:34 VERBOSE[10085]: -- Remote UNIX connection disconnected Apr 7 11:16:38 DEBUG[6513]: Auto destroying call '79c8ee68'