== Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [2155551941@inbound:1] GotoIf("SIP/carrier1-00000008", "0?UNAUTHORIZED,1") in new stack -- Executing [2155551941@inbound:2] Dial("SIP/carrier1-00000008", "SIP/jk-pbx1/2155551941") in new stack == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Audio is at 5060 Video is at 10.1.240.91:5060 Adding codec 0x4 (ulaw) to SDP Adding video codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.19.41:5060: INVITE sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK74572b81 Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.0 Date: Tue, 11 Oct 2011 06:35:42 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 337 v=0 o=root 1185733669 1185733669 IN IP4 10.1.240.91 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.1.240.91 b=CT:384 t=0 0 m=audio 20290 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 24436 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv --- -- Called SIP/jk-pbx1/2155551941 <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK74572b81;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK74572b81;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 1521742333 1521742333 IN IP4 10.0.19.41 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.0.19.41 t=0 0 m=audio 10028 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 99 <-------------> --- (13 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 [Oct 11 02:35:42] WARNING[6013]: chan_sip.c:8740 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 99 Capabilities: us - 0x200004 (ulaw|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.19.41:10028 Peer doesn't provide video list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Transmitting (no NAT) to 10.0.19.41:5060: ACK sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK7fdfefb1 Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.0 Content-Length: 0 --- -- SIP/jk-pbx1-00000009 answered SIP/carrier1-00000008 -- Remotely bridging SIP/carrier1-00000008 and SIP/jk-pbx1-00000009 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.19.41:5060: INVITE sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4037f939 Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 269 v=0 o=root 1185733669 1185733670 IN IP4 10.2.133.151 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.2.133.151 t=0 0 m=audio 49714 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4037f939;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4037f939;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 1521742333 1521742334 IN IP4 10.0.19.41 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.0.19.41 t=0 0 m=audio 10028 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x200004 (ulaw|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.19.41:10028 Peer doesn't provide video set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Transmitting (no NAT) to 10.0.19.41:5060: ACK sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK7f078b6d Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.7.0 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.19.41:5060: INVITE sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4a7ccce5 Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 269 v=0 o=root 1185733669 1185733671 IN IP4 10.2.133.151 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.2.133.151 t=0 0 m=audio 49714 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4a7ccce5;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 104 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK4a7ccce5;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 104 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 1521742333 1521742335 IN IP4 10.0.19.41 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.0.19.41 t=0 0 m=audio 10028 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x200004 (ulaw|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.19.41:10028 Peer doesn't provide video set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Transmitting (no NAT) to 10.0.19.41:5060: ACK sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK316cd50a Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.0.19.41:5060 ---> INVITE sip:8009806858@10.1.240.91:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.19.41:5060;branch=z9hG4bK700e7f5c;rport Max-Forwards: 70 From: ;tag=as1bec550e To: "8009806858" ;tag=as03b11699 Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 268 v=0 o=root 1521742333 1521742336 IN IP4 10.0.19.41 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.0.19.41 t=0 0 m=image 8048 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC <-------------> --- (16 headers 11 lines) --- Sending to 10.0.19.41:5060 (no NAT) Got T.38 offer in SDP in dialog 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 Capabilities: us - 0x200004 (ulaw|h264), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Peer doesn't provide video <--- Transmitting (no NAT) to 10.0.19.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.19.41:5060;branch=z9hG4bK700e7f5c;received=10.0.19.41;rport=5060 From: ;tag=as1bec550e To: "8009806858" ;tag=as03b11699 Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> <--- Reliably Transmitting (no NAT) to 10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.19.41:5060;branch=z9hG4bK700e7f5c;received=10.0.19.41;rport=5060 From: ;tag=as1bec550e To: "8009806858" ;tag=as03b11699 Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 270 v=0 o=root 1185733669 1185733672 IN IP4 10.2.133.151 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.2.133.151 t=0 0 m=image 8026 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:204 a=T38FaxUdpEC:t38UDPFEC <------------> <--- SIP read from UDP:10.0.19.41:5060 ---> ACK sip:8009806858@10.1.240.91:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.19.41:5060;branch=z9hG4bK220e28d8;rport Max-Forwards: 70 From: ;tag=as1bec550e To: "8009806858" ;tag=as03b11699 Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.0 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1185d1254cafe4fb1404acac1bc1f19b@10.0.19.41:5060' Method: OPTIONS set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.19.41:5060: INVITE sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK5ef51095;rport Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1185733669 1185733673 IN IP4 10.1.240.91 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.1.240.91 t=0 0 m=audio 20290 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '51e941e853666082317f7c0275e4e665@10.1.240.91:5060' in 32000 ms (Method: ACK) == Spawn extension (inbound, 2155551941, 2) exited non-zero on 'SIP/carrier1-00000008' <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK5ef51095;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 105 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK5ef51095;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 105 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 1521742333 1521742337 IN IP4 10.0.19.41 s=Asterisk PBX 1.8.7.0 c=IN IP4 10.0.19.41 t=0 0 m=audio 10028 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x200004 (ulaw|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.19.41:10028 Peer doesn't provide video set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Transmitting (no NAT) to 10.0.19.41:5060: ACK sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK696843eb;rport Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Contact: Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.7.0 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.19.41:5060 Reliably Transmitting (no NAT) to 10.0.19.41:5060: BYE sip:2155551941@10.0.19.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK130d87ac;rport Max-Forwards: 70 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 106 BYE User-Agent: Asterisk PBX 1.8.7.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '51e941e853666082317f7c0275e4e665@10.1.240.91:5060' in 32000 ms (Method: ACK) <--- SIP read from UDP:10.0.19.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.240.91:5060;branch=z9hG4bK130d87ac;received=10.1.240.91;rport=5060 From: "8009806858" ;tag=as03b11699 To: ;tag=as1bec550e Call-ID: 51e941e853666082317f7c0275e4e665@10.1.240.91:5060 CSeq: 106 BYE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '51e941e853666082317f7c0275e4e665@10.1.240.91:5060' Method: ACK