Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5e482cbc;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- SIP/uni2-552e answered IAX2/205@205/3 Asterisk*CLI> Retransmitting #3 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK5dd73819;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #4 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK61e8b361;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #5 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes 10 headers, 12 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.100.4:54882 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK646afe20;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b From: ;tag=as723e5f3b To: Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b69a4eb4a98b4746504ce6674a2aa90562b8", response="4933fc9985e0f81cdc1d18aabbe8ae40", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK0547872b;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 106 REGISTER PortaBilling: available-funds:15.72 currency:EUR WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", stale=true Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 10 headers, 0 lines Responding to challenge, registration to domain/host name sip.uni.it 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:sip.uni.it", nonce="4203b7d6c4b8d04c1e0ece422303de227cc0c5d4", response="7b8f300dff6c860a5eb9fe4c27a7b0a7", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK575e275f;rport=5060;received=217.221.92.66 From: ;tag=as723e5f3b To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.794d Call-ID: 2eb141f241b71efb79e2a9e37545e146@127.0.0.1 CSeq: 107 REGISTER PortaBilling: available-funds:15.72 currency:EUR Contact: ;q=0.00;expires=115 Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 Asterisk*CLI> 10 headers, 0 lines Destroying call '2eb141f241b71efb79e2a9e37545e146@127.0.0.1' Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ff0dbb9;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK3ce06ec7;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.101.6 t=0 0 m=audio 18386 RTP/AVP 0 101 c=IN IP4 217.72.101.6 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive 10 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 217.72.101.6:18386 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK103698b5;rport Route: From: "205" ;tag=as1cf84adc To: ;tag=8bc15102 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> Sip read: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport=5060;received=217.221.92.66 From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 8 headers, 0 lines Destroying call '07e6c16356d74ea50968f67e0a025c91@sip.uni.it' Asterisk*CLI> Sip read: 0 headers, 0 lines Asterisk*CLI> exit Executing last minute cleanups ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# cat tuèèypèypeèypèyèypèètyètypètypeètypètyètè typescriopt Script started on Fri 04 Feb 2005 06:54:03 PM CET ]0;root@Asterisk:/usr/src/asterisk[root@Asterisk asterisk]# asterisk -rvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-02/04/05-18:46:21, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-v1-0-02/04/05-18:46:21 currently running on Asterisk (pid = 12098) Asterisk*CLI> Verbosity was 0 and is now 4 Asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 217.72.100.4, port 5060 Reliably Transmitting: BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 (NAT) to 217.72.100.4:5060 == Spawn extension (from-internal, 90102543967, 2) exited non-zero on 'IAX2/205@205/2' -- Executing Macro("IAX2/205@205/2", "hangupcall") in new stack -- Executing ResetCDR("IAX2/205@205/2", "w") in new stack Asterisk*CLI> -- Executing NoCDR("IAX2/205@205/2", "") in new stack -- Executing Wait("IAX2/205@205/2", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/205@205/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/205@205/2' -- Hungup 'IAX2/205@205/2' Asterisk*CLI> -- Accepting AUTHENTICATED call from 213.183.149.139, requested format = 2, actual format = 2 -- Executing SetGroup("IAX2/205@205/3", "205") in new stack -- Executing Dial("IAX2/205@205/3", "SIP/uni2/0102543967||") in new stack We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12099 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 -- Called uni2/0102543967 Asterisk*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport=5060;received=217.221.92.66 From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 INVITE WWW-Authenticate: Digest realm="sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3" Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport From: "205" ;tag=as1cf84adc To: ;tag=a4a48d8b20978897d8e0f5c399e6cc29.7000 Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.72.100.4:5060 We're at 10.0.0.99 port 14868 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:0102543967@sip.uni.it SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport From: "90102543967" ;tag=as1cf84adc To: Contact: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@sip.uni.it", nonce="4203b7cd9f2965ff4859c65bbec3be68f218bfc3", response="0b979e4e8f9fe5c76efc62e825244b13", opaque="" Date: Fri, 04 Feb 2005 17:54:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12099 12100 IN IP4 10.0.0.99 s=session c=IN IP4 10.0.0.99 t=0 0 m=audio 14868 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Asterisk*CLI> a=silenceSupp:off - - - - (NAT) to 217.72.100.4:5060 Asterisk*CLI> -- Registered '205' (AUTHENTICATED) at 213.183.149.139:4569 Asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK2b2b4783;rport=5060;received=217.221.92.66 From: "90102543967" ;tag=as1cf84adc To: Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE PortaBilling: credit-time:53883 currency:EUR Server: Sip EXpress router (0.8.14 (i386/freebsd)) Content-Length: 0 9 headers, 0 lines Asterisk*CLI> Retransmitting #1 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Retransmitting #2 (NAT): BYE sip:0102543967@217.72.100.4:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK73bba698;rport Route: From: "205" ;tag=as09b0e1c5 To: ;tag=e09a8250 Contact: Call-ID: 07e6c16356d74ea50968f67e0a025c91@sip.uni.it CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="390240039504", realm="sip.uni.it", algorithm=MD5, uri="sip:0102543967@217.72.100.4:5061", nonce="4203b72660997fb2d20c18f9881a17070e4908c5", response="9d0cb880a54144c293ac7cda2392abad", opaque="" Content-Length: 0 to 217.72.100.4:5060 Asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Content-Length: 0 8 headers, 0 lines Asterisk*CLI> -- SIP/uni2-552e is ringing Asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.99:5060;rport=5060;received=217.221.92.66;branch=z9hG4bK2b2b4783 To: ;tag=8bc15102 From: "90102543967" ;tag=as1cf84adc Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it CSeq: 103 INVITE Record-Route: Contact: Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 4916 6679 IN IP4 217.72.101.6 s=SIP Call c=IN IP4 217.72.100.4 t=0 0 m=audio 54882 RTP/AVP 0 101 c=IN IP4 217.72.100.4 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000