localhost*CLI> Verbosity was 16 and is now 19 localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> Really destroying SIP dialog 'MjJmN2YyY2U3YzI5NDYyZjk5MzM2OTNmZjBiY2RlOWY.' Method: REGISTER localhost*CLI> sip set debug on SIP Debugging re-enabled localhost*CLI> localhost*CLI> localhost*CLI> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> localhost*CLI> sip set history on SIP History Recording Enabled (use 'sip show history') localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> == Manager 'corelis' logged on from 172.30.0.117 <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.0.55:4597: INVITE sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6fb6247f;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-rc2 Date: Tue, 08 Feb 2011 06:14:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 v=0 o=root 1264113285 1264113285 IN IP4 172.30.0.117 s=Asterisk PBX 1.8.3-rc2 c=IN IP4 172.30.0.117 t=0 0 m=audio 17536 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 --- <--- SIP read from UDP:172.30.0.55:4597 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6fb6247f;rport=5060 Contact: To: ;tag=9616ce2b From: "20000";tag=as2bda31cf Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.30.0.55:4597 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6fb6247f;rport=5060 Contact: To: ;tag=9616ce2b From: "20000";tag=as2bda31cf Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 206 v=0 o=3cxVCE 57330180 210074865 IN IP4 172.30.0.55 s=3cxVCE Audio Call c=IN IP4 172.30.0.55 t=0 0 m=audio 40000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 9 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 - 0x4 (ulaw), 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 172.30.0.55:40000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Transmitting (NAT) to 172.30.0.55:4597: ACK sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK3a90ff40;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3-rc2 Content-Length: 0 --- > Channel SIP/20000-00000018 was answered. -- Executing [20001@tenant01:1] Dial("SIP/20000-00000018", "SIP/20001,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.1.48:2144: INVITE sip:20001@172.30.1.48:2144;rinstance=fdd16cfde13f9074 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6f02d518;rport Max-Forwards: 70 From: "Sip" ;tag=as5f5561cb To: Contact: Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-rc2 Date: Tue, 08 Feb 2011 06:14:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 v=0 o=root 1047713130 1047713130 IN IP4 172.30.0.117 s=Asterisk PBX 1.8.3-rc2 c=IN IP4 172.30.0.117 t=0 0 m=audio 11846 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 --- -- Called 20001 <--- SIP read from UDP:172.30.1.48:2144 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6f02d518;rport=5060 Contact: To: ;tag=69365c4e From: "Sip";tag=as5f5561cb Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/20001-00000019 is ringing <--- SIP read from UDP:172.30.1.48:2144 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6f02d518;rport=5060 Contact: To: ;tag=69365c4e From: "Sip";tag=as5f5561cb Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO Content-Type: application/sdp upported: replaces User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 207 v=0 o=3cxVCE 104574495 188742705 IN IP4 172.30.1.48 s=3cxVCE Audio Call c=IN IP4 172.30.1.48 t=0 0 m=audio 40006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 9 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 - 0x4 (ulaw), 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 172.30.1.48:40006 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.1.48:2144 Transmitting (NAT) to 172.30.1.48:2144: ACK sip:20001@172.30.1.48:2144;rinstance=fdd16cfde13f9074 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK3f2c7d76;rport Max-Forwards: 70 From: "Sip" ;tag=as5f5561cb To: ;tag=69365c4e Contact: Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3-rc2 Content-Length: 0 --- -- SIP/20001-00000019 answered SIP/20000-00000018 -- Remotely bridging SIP/20000-00000018 and SIP/20001-00000019 set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.0.55:4597: INVITE sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK575e7fa0;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3-rc2 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: 236 v=0 o=root 1264113285 1264113286 IN IP4 172.30.1.48 s=Asterisk PBX 1.8.3-rc2 c=IN IP4 172.30.1.48 t=0 0 m=audio 40006 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 --- set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.1.48:2144 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.1.48:2144: INVITE sip:20001@172.30.1.48:2144;rinstance=fdd16cfde13f9074 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK49c93d59;rport Max-Forwards: 70 From: "Sip" ;tag=as5f5561cb To: ;tag=69365c4e Contact: Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3-rc2 llow: 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: 236 v=0 o=root 1047713130 1047713131 IN IP4 172.30.0.55 s=Asterisk PBX 1.8.3-rc2 c=IN IP4 172.30.0.55 t=0 0 m=audio 40000 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 --- <--- SIP read from UDP:172.30.0.55:4597 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK575e7fa0;rport=5060 Contact: To: ;tag=9616ce2b From: "20000";tag=as2bda31cf Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 206 v=0 o=3cxVCE 57330180 210074866 IN IP4 172.30.0.55 s=3cxVCE Audio Call c=IN IP4 172.30.0.55 t=0 0 m=audio 40000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 9 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 - 0x4 (ulaw), 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 172.30.0.55:40000 set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Transmitting (NAT) to 172.30.0.55:4597: ACK sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK421853a0;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3-rc2 Content-Length: 0 --- <--- SIP read from UDP:172.30.1.48:2144 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK49c93d59;rport=5060 Contact: To: ;tag=69365c4e From: "Sip";tag=as5f5561cb Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO Content-Type: application/sdp upported: replaces User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 207 v=0 o=3cxVCE 104574495 188742706 IN IP4 172.30.1.48 s=3cxVCE Audio Call c=IN IP4 172.30.1.48 t=0 0 m=audio 40006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 9 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 - 0x4 (ulaw), 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 172.30.1.48:40006 set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.1.48:2144 Transmitting (NAT) to 172.30.1.48:2144: ACK sip:20001@172.30.1.48:2144;rinstance=fdd16cfde13f9074 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6df4c11e;rport Max-Forwards: 70 From: "Sip" ;tag=as5f5561cb To: ;tag=69365c4e Contact: Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3-rc2 Content-Length: 0 --- <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> localhost*CLI> core show chan channel channels channeltypes channeltype localhost*CLI> core show channels Channel Location State Application(Data) SIP/20001-00000019 (None) Up AppDial((Outgoing Line)) SIP/20000-00000018 20001@tenant01:1 Up Dial(SIP/20001,60) 2 active channels 1 active call 20 calls processed <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.0.55:4597: INVITE sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6d4d9b79;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.3-rc2 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: 238 v=0 o=root 1264113285 1264113287 IN IP4 172.30.0.117 s=Asterisk PBX 1.8.3-rc2 c=IN IP4 172.30.0.117 t=0 0 m=audio 17536 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 --- [Feb 8 15:15:47] WARNING[4424]: rtp_engine.c:1209 remote_bridge_loop: Channel 'AsyncGoto/SIP/20001-00000019' failed to break RTP bridge == Spawn extension (tenant01, 9060000, 1) exited non-zero on 'SIP/20000-00000018' -- Executing [9060000@tenant01:1] Set("SIP/20000-00000018", "Routing=start") in new stack -- Executing [9060000@tenant01:2] Queue("SIP/20000-00000018", "9060000") in new stack -- Started music on hold, class 'default', on SIP/20000-00000018 -- Stopped music on hold on SIP/20000-00000018 == Spawn extension (tenant01, 9060000, 2) exited non-zero on 'SIP/20000-00000018' Scheduling destruction of SIP dialog '5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060' in 32000 ms (Method: INVITE) -- Executing [9060000@tenant01:1] Set("SIP/20001-00000019", "Routing=start") in new stack -- Executing [9060000@tenant01:2] Queue("SIP/20001-00000019", "9060000") in new stack -- Started music on hold, class 'default', on SIP/20001-00000019 <--- SIP read from UDP:172.30.0.55:4597 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK6d4d9b79;rport=5060 Contact: To: ;tag=9616ce2b From: "20000";tag=as2bda31cf Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 206 v=0 o=3cxVCE 57330180 210074867 IN IP4 172.30.0.55 s=3cxVCE Audio Call c=IN IP4 172.30.0.55 t=0 0 m=audio 40000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 9 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 - 0x4 (ulaw), 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 172.30.0.55:40000 set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Transmitting (NAT) to 172.30.0.55:4597: ACK sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK5f7ed973;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Contact: Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.3-rc2 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.55:4597 Reliably Transmitting (NAT) to 172.30.0.55:4597: BYE sip:20000@172.30.0.55:4597;rinstance=dcef5530dfd4dcb8 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK40d28ed7;rport Max-Forwards: 70 From: "20000" ;tag=as2bda31cf To: ;tag=9616ce2b Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 1.8.3-rc2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:172.30.0.55:4597 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.117:5060;branch=z9hG4bK40d28ed7;rport=5060 Contact: To: ;tag=9616ce2b From: "20000";tag=as2bda31cf Call-ID: 5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060 CSeq: 105 BYE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '5b8a69a845f2cf21165142734884f56c@172.30.0.117:5060' Method: INVITE <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> BYE sip:20000@172.30.0.117:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.48:2144;branch=z9hG4bK-d8754z-80762b04695fa849-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Sip";tag=as5f5561cb From: ;tag=69365c4e Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 2 BYE User-Agent: 3CXPhone 4.0.10858.0 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 172.30.1.48:2144 (NAT) Scheduling destruction of SIP dialog '1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 172.30.1.48:2144 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.48:2144;branch=z9hG4bK-d8754z-80762b04695fa849-1---d8754z-;received=172.30.1.48;rport=2144 From: ;tag=69365c4e To: "Sip";tag=as5f5561cb Call-ID: 1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060 CSeq: 2 BYE Server: Asterisk PBX 1.8.3-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> -- Stopped music on hold on SIP/20001-00000019 == Spawn extension (tenant01, 9060000, 2) exited non-zero on 'SIP/20001-00000019' <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> Really destroying SIP dialog '1db5ada3796abcb62b4779b825be2d5b@172.30.0.117:5060' Method: BYE <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> localhost*CLI> localhost*CLI> localhost*CLI> <--- SIP read from UDP:172.30.1.48:2144 ---> <-------------> <--- SIP read from UDP:172.30.0.55:4597 ---> <-------------> localhost*CLI>