*CLI> sip debug SIP Debugging enabled *CLI> <-- SIP read from 88.149.128.60:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK4f5e6239;received=81.75.230.211;rport=5060 From: "055451360" ;tag=as51ef5b15 To: ;tag=as2fea9748 Call-ID: 0b0abf1766819e1c6d27c9d91e6359f9@squillo.it CSeq: 102 INVITE User-Agent: www.squillo.it Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- <-- SIP read from 88.149.128.60:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK4f5e6239;received=81.75.230.211;rport=5060 From: "055451360" ;tag=as51ef5b15 To: ;tag=as2fea9748 Call-ID: 0b0abf1766819e1c6d27c9d91e6359f9@squillo.it CSeq: 102 INVITE User-Agent: www.squillo.it Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (NAT) to 88.149.168.208:5060: OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport From: "asterisk" ;tag=as7f51e0b2 To: Contact: Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 88.149.168.208:5060: OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport From: "asterisk" ;tag=as79bf5fc0 To: Contact: Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 88.149.128.60:5060: INVITE sip:s@81.75.230.211 SIP/2.0 Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;rport From: "055451360" ;tag=as684ecc32 To: Contact: Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 INVITE User-Agent: www.squillo.it Date: Thu, 29 Mar 2007 21:10:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 341 v=0 o=root 25171 25171 IN IP4 88.149.128.60 s=session c=IN IP4 88.149.128.60 t=0 0 m=audio 32664 RTP/AVP 8 4 18 97 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 15 lines) --- Using INVITE request as basis request - 0806df6f1d5305df37317c8332aad052@squillo.it Sending to 88.149.128.60 : 5060 (NAT) Found peer 'squillo' Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 88.149.128.60:32664 Found description format PCMA Found description format G723 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format telephone-event Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x50f (g723|gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in default (domain 81.75.230.211) list_route: hop: Transmitting (NAT) to 88.149.128.60:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=88.149.128.60;rport=5060 From: "055451360" ;tag=as684ecc32 To: Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Answer("SIP/criolo-08157860", "") in new stack We're at 81.75.230.211 port 11916 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 88.149.128.60:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=88.149.128.60;rport=5060 From: "055451360" ;tag=as684ecc32 To: ;tag=as29a617c0 Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 15009 15009 IN IP4 81.75.230.211 s=session c=IN IP4 81.75.230.211 t=0 0 m=audio 11916 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Dial("SIP/criolo-08157860", "SIP/7199") in new stack Destroying call '389bf50f75ed13260163c22d1a944b01@192.168.0.2' Mar 29 23:10:51 NOTICE[15035]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup("SIP/criolo-08157860", "") in new stack == Spawn extension (default, s, 3) exited non-zero on 'SIP/criolo-08157860' Scheduling destruction of call '0806df6f1d5305df37317c8332aad052@squillo.it' in 32000 ms <-- SIP read from 88.149.128.60:5060: ACK sip:s@81.75.230.211 SIP/2.0 Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK16442e1d;rport From: "055451360" ;tag=as684ecc32 To: ;tag=as29a617c0 Contact: Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 ACK User-Agent: www.squillo.it Content-Length: 0 --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 88.149.128.60, port 5060 Reliably Transmitting (NAT) to 88.149.128.60:5060: CANCEL :s@81.75.230.211 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK19e8983e;rport From: To: "055451360" ;tag=as684ecc32 Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '0806df6f1d5305df37317c8332aad052@squillo.it' in 32000 ms Retransmitting #1 (NAT) to 88.149.168.208:5060: OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport From: "asterisk" ;tag=as7f51e0b2 To: Contact: Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #1 (NAT) to 88.149.168.208:5060: OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport From: "asterisk" ;tag=as79bf5fc0 To: Contact: Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 88.149.128.60:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=81.75.230.211;rport=5060 From: "055451360" ;tag=as684ecc32 To: ;tag=as29a617c0 Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 INVITE User-Agent: www.squillo.it Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- <-- SIP read from 88.149.128.60:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK19e8983e;received=81.75.230.211;rport=5060 From: To: "055451360" ;tag=as684ecc32 Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 101 CANCEL User-Agent: www.squillo.it Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- Retransmitting #2 (NAT) to 88.149.168.208:5060: OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport From: "asterisk" ;tag=as7f51e0b2 To: Contact: Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #2 (NAT) to 88.149.168.208:5060: OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport From: "asterisk" ;tag=as79bf5fc0 To: Contact: Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #3 (NAT) to 88.149.168.208:5060: OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport From: "asterisk" ;tag=as7f51e0b2 To: Contact: Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #3 (NAT) to 88.149.168.208:5060: OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport From: "asterisk" ;tag=as79bf5fc0 To: Contact: Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 88.149.128.60:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=81.75.230.211;rport=5060 From: "055451360" ;tag=as684ecc32 To: ;tag=as29a617c0 Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it CSeq: 102 INVITE User-Agent: www.squillo.it Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines) --- Retransmitting #4 (NAT) to 88.149.168.208:5060: OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport From: "asterisk" ;tag=as7f51e0b2 To: Contact: Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '79037f1f2d075175711a0c47476ff400@81.75.230.211' Retransmitting #4 (NAT) to 88.149.168.208:5060: OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport From: "asterisk" ;tag=as79bf5fc0 To: Contact: Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 21:10:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '0b3e5ee676d6085e7c43412133d28652@81.75.230.211' Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (2).