sip*CLI> sip set debug peer trunk_3 SIP Debugging Enabled for IP: 81.23.228.150:5060 sip*CLI> <--- SIP read from 81.23.228.150:5060 ---> INVITE sip:s@217.195.248.252 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 From: "0651455735" ;tag=as4bf5f515 To: Contact: Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 04 Jul 2007 14:15:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 392 v=0 o=root 16089 16089 IN IP4 193.67.155.227 s=session c=IN IP4 83.149.75.105 t=0 0 m=audio 62650 RTP/AVP 3 18 5 8 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:5 DVI4/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (15 headers 17 lines) --- Sending to 81.23.228.150 : 5060 (no NAT) Using INVITE request as basis request - 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl Found peer 'trunk_3' Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 110 Found RTP audio format 101 Peer audio RTP is at port 83.149.75.105:62650 Found description format GSM for ID 3 Found description format G729 for ID 18 Found description format DVI4 for ID 5 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format iLBC for ID 97 Found description format speex for ID 110 Found description format telephone-event for ID 101 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)/video=0x0 (nothing), combined - 0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 83.149.75.105:62650 Peer video RTP is at port 83.149.75.105:5128 Looking for s in DID_trunk_3 (domain 217.195.248.252) list_route: hop: <--- Transmitting (no NAT) to 81.23.228.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 From: "0651455735" ;tag=as4bf5f515 To: Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [s@DID_trunk_3:1] Goto("SIP/31105117619-b4608d48", "default|102|1") in new stack -- Goto (default,102,1) -- Executing [102@default:1] Macro("SIP/31105117619-b4608d48", "stdexten|102|SIP/102") in new stack -- Executing [s@macro-stdexten:1] Dial("SIP/31105117619-b4608d48", "SIP/102|20") in new stack -- Called 102 -- SIP/102-09434fc8 is ringing <--- Transmitting (no NAT) to 81.23.228.150:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 From: "0651455735" ;tag=as4bf5f515 To: ;tag=as62fba3c0 Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> <--- SIP read from 81.23.228.150:5060 ---> CANCEL sip:s@217.195.248.252 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 From: "0651455735" ;tag=as4bf5f515 To: Contact: Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 81.23.228.150 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 81.23.228.150:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 From: "0651455735" ;tag=as4bf5f515 To: ;tag=as62fba3c0 Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 81.23.228.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150 Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060 Record-Route: From: "0651455735" ;tag=as4bf5f515 To: ;tag=as62fba3c0 Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl CSeq: 102 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/31105117619-b4608d48' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/31105117619-b4608d48' sip*CLI> <--- SIP read from 81.23.228.150:5060 ---> ACK sip:s@217.195.248.252 SIP/2.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0 From: "0651455735" ;tag=as4bf5f515 Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl To: ;tag=as62fba3c0 CSeq: 102 ACK User-Agent: OpenSer (1.1.0-tls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl' Method: ACK RTP-stats * Our Receiver: SSRC: 0 Received packets: 0 Lost packets: 0 Jitter: 0.0000 Transit: 0.0000 RR-count: 0 * Our Sender: SSRC: 998330856 Sent packets: 0 Lost packets: 0 Jitter: 0 SR-count: 0 RTT: 0.000000 RTP-stats * Our Receiver: SSRC: 0 Received packets: 0 Lost packets: 0 Jitter: 0.0000 Transit: 0.0000 RR-count: 0 * Our Sender: SSRC: 463146790 Sent packets: 0 Lost packets: 0 Jitter: 0 SR-count: 0 RTT: 0.000000 RTP-stats * Our Receiver: SSRC: 0 Received packets: 0 Lost packets: 0 Jitter: 0.0000 Transit: 0.0000 RR-count: 0 * Our Sender: SSRC: 1504204863 Sent packets: 0 Lost packets: 0 Jitter: 0 SR-count: 0 RTT: 0.000000 RTP-stats * Our Receiver: SSRC: 0 Received packets: 0 Lost packets: 0 Jitter: 0.0000 Transit: 0.0000 RR-count: 0 * Our Sender: SSRC: 2088594371 Sent packets: 0 Lost packets: 0 Jitter: 0 SR-count: 0 RTT: 0.000000