emigreer*CLI> <--- SIP read from 87.249.114.95:5060 ---> INVITE sip:31880016403@alt001.com SIP/2.0 Record-Route: Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK280b.15137a4.0 Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK2b207520;rport=8066 From: " " ;tag=as65cf3b60 To: Contact: Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: SpeakUp PSTN Max-Forwards: 69 Date: Fri, 28 Mar 2008 10:26:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 23936 23936 IN IP4 217.114.103.72 s=session c=IN IP4 217.114.103.72 t=0 0 m=audio 14492 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (15 headers 12 lines) --- Sending to 87.249.114.95 : 5060 (no NAT) Using INVITE request as basis request - 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 Found peer 'speakup01' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 217.114.103.72:14492 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 217.114.103.72:14492 Looking for 31880016403 in incoming (domain alt001.com) list_route: hop: emigreer*CLI> <--- Transmitting (no NAT) to 87.249.114.95:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK280b.15137a4.0;received=87.249.114.95 Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK2b207520;rport=8066 Record-Route: From: " " ;tag=as65cf3b60 To: Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [31880016403@incoming:1] Set("SIP/alt001-sip-081cd650", "CUSTOMER_EXT=SIP/test2/31880016403") in new stack -- Executing [31880016403@incoming:2] Goto("SIP/alt001-sip-081cd650", "dial-customer|31880016403|1") in new stack -- Goto (dial-customer,31880016403,1) -- Executing [31880016403@dial-customer:1] Log("SIP/alt001-sip-081cd650", "NOTICE|Call-id:1A-1206699991.970 Dial-Customer From:31641013814 To:31880016403 Using:"SIP/test2/31880016403"") in new stack [Mar 28 11:26:31] NOTICE[30914]: Ext. 31880016403:1 @ dial-customer: Call-id:1A-1206699991.970 Dial-Customer From:31641013814 To:31880016403 Using:"SIP/test2/31880016403" -- Executing [31880016403@dial-customer:2] Verbose("SIP/alt001-sip-081cd650", "1|--- Now:11:26:31 Call-id:1A-1206699991.970 --> Call From:-31641013814-" " To:"31880016403" Using:"SIP/test2/31880016403" ---") in new stack --- Now:11:26:31 Call-id:1A-1206699991.970 --> Call From:-31641013814-" " To:"31880016403" Using:"SIP/test2/31880016403" --- -- Executing [31880016403@dial-customer:3] Dial("SIP/alt001-sip-081cd650", "SIP/test2/31880016403") in new stack emigreer*CLI> Audio is at 87.253.148.78 port 13036 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 92.65.24.219:5060: INVITE sip:31880016403@92.65.24.219 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK2bfe8a40;rport From: " " ;tag=as520b2f51 To: Contact: Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 INVITE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Date: Fri, 28 Mar 2008 10:26:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 13894 13894 IN IP4 217.114.103.72 s=session c=IN IP4 217.114.103.72 t=0 0 m=audio 14492 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called test2/31880016403 emigreer*CLI> <--- SIP read from 92.65.24.219:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK2bfe8a40;received=87.253.148.78;rport=5060 From: " " ;tag=as520b2f51 To: Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- emigreer*CLI> <--- SIP read from 92.65.24.219:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK2bfe8a40;received=87.253.148.78;rport=5060 From: " " ;tag=as520b2f51 To: ;tag=as337c0952 Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- SIP/test2-081bbd60 is ringing <--- Transmitting (no NAT) to 87.249.114.95:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK280b.15137a4.0;received=87.249.114.95 Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK2b207520;rport=8066 Record-Route: From: " " ;tag=as65cf3b60 To: ;tag=as494afd48 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> emigreer*CLI> emigreer*CLI> <--- SIP read from 92.65.24.219:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK2bfe8a40;received=87.253.148.78;rport=5060 From: " " ;tag=as520b2f51 To: ;tag=as337c0952 Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 12217 12217 IN IP4 92.65.24.219 s=session c=IN IP4 92.65.24.219 t=0 0 m=audio 12710 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (12 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 92.65.24.219:12710 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xa (gsm|alaw), peer - audio=0xa (gsm|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 92.65.24.219:12710 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 92.65.24.219, port 5060 Transmitting (no NAT) to 92.65.24.219:5060: ACK sip:31880016403@92.65.24.219 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK46dab06e;rport From: " " ;tag=as520b2f51 To: ;tag=as337c0952 Contact: Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 ACK User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- -- SIP/test2-081bbd60 answered SIP/alt001-sip-081cd650 Audio is at 87.253.148.78 port 12730 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 87.249.114.95:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK280b.15137a4.0;received=87.249.114.95 Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK2b207520;rport=8066 Record-Route: From: " " ;tag=as65cf3b60 To: ;tag=as494afd48 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 13894 13894 IN IP4 92.65.24.219 s=session c=IN IP4 92.65.24.219 t=0 0 m=audio 12710 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/alt001-sip-081cd650 and SIP/test2-081bbd60 emigreer*CLI> <--- SIP read from 87.249.114.95:5060 ---> ACK sip:31880016403@87.253.148.78 SIP/2.0 Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK280b.15137a4.2 Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK62b91301;rport=8066 From: " " ;tag=as65cf3b60 To: ;tag=as494afd48 Contact: Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 ACK User-Agent: SpeakUp PSTN Max-Forwards: 69 ontent-Length: 0 P-hint: RR-enforced <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '382e45e0497b0f9550e32fcb62cbc32a@87.253.148.78' Method: OPTIONS emigreer*CLI> <--- SIP read from 92.65.24.219:5060 ---> BYE sip:31641013814@87.253.148.78 SIP/2.0 Via: SIP/2.0/UDP 92.65.24.219:5060;branch=z9hG4bK2c88b16a;rport From: ;tag=as337c0952 To: " " ;tag=as520b2f51 Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 92.65.24.219 : 5060 (NAT) emigreer*CLI> <--- Transmitting (NAT) to 92.65.24.219:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.65.24.219:5060;branch=z9hG4bK2c88b16a;received=92.65.24.219;rport=5060 From: ;tag=as337c0952 To: " " ;tag=as520b2f51 Call-ID: 7ee4ae924709430802cc33447f54df75@87.253.148.78 CSeq: 102 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 87.249.114.95, port 5060 Audio is at 87.253.148.78 port 12730 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 87.249.114.95:5060: INVITE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK3354f5c6;rport Route: From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Contact: Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 13894 13895 IN IP4 87.253.148.78 s=session c=IN IP4 87.253.148.78 t=0 0 m=audio 12730 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Spawn extension (dial-customer, 31880016403, 3) exited non-zero on 'SIP/alt001-sip-081cd650' -- Executing [h@dial-customer:1] Verbose("SIP/alt001-sip-081cd650", "1|--- Now:11:26:37 Call-id:1A-1206699991.970 --> Call From:"31641013814" To:"SIP/test2/31880016403" lasted 6 seconds. (ANSWER) ---") in new stack --- Now:11:26:37 Call-id:1A-1206699991.970 --> Call From:"31641013814" To:"SIP/test2/31880016403" lasted 6 seconds. (ANSWER) --- -- Executing [h@dial-customer:2] Log("SIP/alt001-sip-081cd650", "NOTICE|Call-id:1A-1206699991.970 Dial-Customer Using:"SIP/test2/31880016403" Status:ANSWER Duration:6 sec.") in new stack [Mar 28 11:26:37] NOTICE[30914]: Ext. h:2 @ dial-customer: Call-id:1A-1206699991.970 Dial-Customer Using:"SIP/test2/31880016403" Status:ANSWER Duration:6 sec. -- Executing [h@dial-customer:3] Hangup("SIP/alt001-sip-081cd650", "16") in new stack == Spawn extension (dial-customer, h, 3) exited non-zero on 'SIP/alt001-sip-081cd650' emigreer*CLI> <--- SIP read from 87.249.114.95:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK3354f5c6;rport=5060 From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: SpeakUp PSTN Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 23936 23937 IN IP4 217.114.103.72 s=session c=IN IP4 217.114.103.72 t=0 0 m=audio 14492 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (12 headers 12 lines) --- Scheduling destruction of SIP dialog '7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72' in 6400 ms (Method: ACK) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 217.114.103.72:14492 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xa (gsm|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 217.114.103.72:14492 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 217.114.103.72, port 8066 Transmitting (no NAT) to 217.114.103.72:8066: ACK sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK45b815e8;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Contact: Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 ACK User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 217.114.103.72, port 8066 Reliably Transmitting (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72' in 6400 ms (Method: ACK) Really destroying SIP dialog '7ee4ae924709430802cc33447f54df75@87.253.148.78' Method: BYE emigreer*CLI> <--- SIP read from 87.249.114.95:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK3354f5c6;rport=5060 From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: SpeakUp PSTN Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 23936 23937 IN IP4 217.114.103.72 s=session c=IN IP4 217.114.103.72 t=0 0 m=audio 14492 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (12 headers 12 lines) --- Retransmitting #1 (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- Retransmitting #2 (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- Retransmitting #3 (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- Retransmitting #4 (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0 --- emigreer*CLI> <--- SIP read from 87.249.114.95:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK3354f5c6;rport=5060 From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 102 INVITE User-Agent: SpeakUp PSTN Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 23936 23937 IN IP4 217.114.103.72 s=session c=IN IP4 217.114.103.72 t=0 0 m=audio 14492 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (12 headers 12 lines) --- Retransmitting #5 (no NAT) to 217.114.103.72:8066: BYE sip:31641013814@217.114.103.72:8066 SIP/2.0 Via: SIP/2.0/UDP 87.253.148.78:5060;branch=z9hG4bK70feb3ea;rport From: ;tag=as494afd48 To: " " ;tag=as65cf3b60 Call-ID: 7660ab6c6ee902ea5ae9523476f080c0@217.114.103.72 CSeq: 103 BYE User-Agent: Asterisk 1.4 on emigreer.bokxing.nl Max-Forwards: 70 Content-Length: 0