rendered paste bodyPP1 asterisk # asterisk -rAsterisk 1.2.14, Copyright (C) 1999 - 2006 Digium, Inc. and others.Created by Mark Spencer <markster@digium.com>fAsterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it undercertain conditions. Type 'show license' for details.=========================================================================Connected to Asterisk 1.2.14 currently running on PP1 (pid = 4971) -- Remote UNIX connectionVerbosity is at least 3PP1*CLI> set verbose 11Verbosity was 3 and is now 11PP1*CLI> sip debugSIP Debugging enabled12 headers, 0 linesReliably Transmitting (NAT) to 24.7.133.237:1204:OPTIONS sip:541@24.7.133.237:1204 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5ff8956e;rportFrom: "Unknown" <sip:Unknown@38.103.144.10>;tag=as750f4b04To: <sip:541@24.7.133.237:1204>Contact: <sip:Unknown@38.103.144.10>Call-ID: 1f13bfab700cf154150aed5040b81fcd@38.103.144.10CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Wed, 14 Feb 2007 10:45:05 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0---REGISTER 12 headers, 0 linesReliably Transmitting (NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK78474ef7;rportFrom: <sip:9048664245@4.78.224.161>;tag=as71f849e3To: <sip:9048664245@4.78.224.161>Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1CSeq: 108 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 200 OKFrom: <sip:9048664245@4.78.224.161>;tag=as71f849e3To: <sip:9048664245@4.78.224.161>;tag=192.168.67.2+1+0+a67f3e5fServer: DC-SIP/2.0Organization:Content-Length: 0Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1CSeq: 108 REGISTERVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK78474ef7Contact: sip:s@4.78.224.161:5060;expires=3600--- (10 headers 0 lines) ---Scheduling destruction of call '69555ff937282717319589672fb5bae1@127.0.0.1' in 32000 msPP1*CLI><-- SIP read from 24.7.133.237:1204:SIP/2.0 200 OKVia: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5ff8956e;rport=5060From: "Unknown" <sip:Unknown@38.103.144.10>;tag=as750f4b04To: <sip:541@24.7.133.237:1204>;tag=000c753543badb11abbe005056c00008Call-ID: 1f13bfab700cf154150aed5040b81fcd@38.103.144.10CSeq: 102 OPTIONSContact: <sip:541@24.7.133.237:1204>Max-Forwards: 70Server: SIPPER for PhonerLiteAllow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFYContent-Length: 0--- (11 headers 0 lines) ---Destroying call '1f13bfab700cf154150aed5040b81fcd@38.103.144.10'REGISTER 12 headers, 0 linesReliably Transmitting (NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK202c65f2;rportFrom: <sip:9048664204@4.78.224.161>;tag=as7a9b39c9To: <sip:9048664204@4.78.224.161>Call-ID: 766a1e4d3177ae566ecf85126063aef6@127.0.0.1CSeq: 108 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 409 ConflictVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK202c65f2;rportFrom: <sip:9048664204@192.168.67.2>;tag=as7a9b39c9To: <sip:9048664204@192.168.67.2>Call-ID: 766a1e4d3177ae566ecf85126063aef6@127.0.0.1CSeq: 108 REGISTERContent-Length: 0--- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161Destroying call '766a1e4d3177ae566ecf85126063aef6@127.0.0.1'12 headers, 0 linesReliably Transmitting (NAT) to 68.218.191.131:50684:OPTIONS sip:543@68.218.191.131:50684;rinstance=654b8b1488039a8f SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4b8aad6d;rportFrom: "Unknown" <sip:Unknown@38.103.144.10>;tag=as11f012aaTo: <sip:543@68.218.191.131:50684;rinstance=654b8b1488039a8f>Contact: <sip:Unknown@38.103.144.10>Call-ID: 73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Wed, 14 Feb 2007 10:45:06 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0---REGISTER 12 headers, 0 linesReliably Transmitting (NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK6fa656c7;rportFrom: <sip:9048664201@4.78.224.161>;tag=as4f0b7dd1To: <sip:9048664201@4.78.224.161>Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1CSeq: 108 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 409 ConflictVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK6fa656c7;rportFrom: <sip:9048664201@192.168.67.2>;tag=as4f0b7dd1To: <sip:9048664201@192.168.67.2>Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1CSeq: 108 REGISTERContent-Length: 0--- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161Destroying call '54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1'PP1*CLI><-- SIP read from 68.218.191.131:50684:SIP/2.0 200 OKVia: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4b8aad6d;rport=5060Contact: <sip:192.168.15.101:35404>To: <sip:543@68.218.191.131:50684;rinstance=654b8b1488039a8f>;tag=072bd245From: "Unknown"<sip:Unknown@38.103.144.10>;tag=as11f012aaCall-ID: 73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10CSeq: 102 OPTIONSAccept: application/sdpAccept-Language: enAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFOUser-Agent: X-Lite release 1006e stamp 34025Content-Length: 0--- (12 headers 0 lines) ---Destroying call '73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10'PP1*CLI><-- SIP read from 4.78.224.161:5060:INVITE sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69Session-Expires: 3600;Refresher=uacSupported: timerTo: <sip:s@4.78.224.161:5060>From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dCall-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fcecedContact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Type: application/sdpContent-Length: 144v=0o=dllsl3-ntsbc02 2147483647 2147483647 IN IP4 4.78.224.161s=sip callc=IN IP4 4.78.224.161t=0 0m=audio 30064 RTP/AVP 0a=ptime:20--- (16 headers 7 lines) ---Using INVITE request as basis request - 415219-3380409986-455979@dllsl3-ntsbc02.une3.comSending to 4.78.224.161 : 5060 (NAT)Found peer 'suntel-4245'Found RTP audio format 0Peer audio RTP is at port 4.78.224.161:30064Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)Looking for s in from-trunk (domain 4.78.224.161)list_route: hop: <sip:anonymous@4.78.224.161:5060>Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fceced;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dTo: <sip:s@4.78.224.161:5060>Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0--- -- Executing Set("SIP/9048664245-b682d660", "FROM_DID=s") in new stack -- Executing Goto("SIP/9048664245-b682d660", "ivr-3|s|1") in new stack -- Goto (ivr-3,s,1) -- Executing Set("SIP/9048664245-b682d660", "LOOPCOUNT=0") in new stack -- Executing Set("SIP/9048664245-b682d660", "__DIR-CONTEXT=zonemessages") in new stack -- Executing Answer("SIP/9048664245-b682d660", "") in new stackWe're at 38.103.144.10 port 12284Adding codec 0x4 (ulaw) to SDPReliably Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fceced;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dTo: <sip:s@4.78.224.161:5060>;tag=as3b456056Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Type: application/sdpContent-Length: 160v=0o=root 4971 4971 IN IP4 38.103.144.10s=sessionc=IN IP4 38.103.144.10t=0 0m=audio 12284 RTP/AVP 0a=rtpmap:0 PCMU/8000a=silenceSupp:off - - - ---- -- Executing Wait("SIP/9048664245-b682d660", "1") in new stackPP1*CLI><-- SIP read from 4.78.224.161:5060:ACK sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69To: <sip:s@4.78.224.161:5060>;tag=as3b456056From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dCall-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 1 ACKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKde9947c8913a6b421120479bf99d0b64Contact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Length: 0--- (13 headers 0 lines) --- -- Executing Set("SIP/9048664245-b682d660", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing Set("SIP/9048664245-b682d660", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/9048664245-b682d660", "custom/SpanishEnterYourPinSunTel") in new stack -- Playing 'custom/SpanishEnterYourPinSunTel' (language 'en')PP1*CLI><-- SIP read from 4.78.224.161:5060:BYE sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69To: <sip:s@4.78.224.161:5060>;tag=as3b456056From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dCall-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 2 BYEVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKad1d6fb96fe2a550b78211c41adfa17aContact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Length: 0--- (13 headers 0 lines) ---Sending to 4.78.224.161 : 5060 (NAT)Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKad1d6fb96fe2a550b78211c41adfa17a;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+13d854+7f64318dTo: <sip:s@4.78.224.161:5060>;tag=as3b456056Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.comCSeq: 2 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0--- == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/9048664245-b682d660' -- Executing Hangup("SIP/9048664245-b682d660", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9048664245-b682d660'Destroying call '415219-3380409986-455979@dllsl3-ntsbc02.une3.com'PP1*CLI>PP1 asterisk # asterisk -rAsterisk 1.2.14, Copyright (C) 1999 - 2006 Digium, Inc. and others.Created by Mark Spencer <markster@digium.com>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it undercertain conditions. Type 'show license' for details.=========================================================================Connected to Asterisk 1.2.14 currently running on PP1 (pid = 4971) -- Remote UNIX connectionVerbosity is at least 11Destroying call '766a1e4d3177ae566ecf85126063aef6@127.0.0.1'REGISTER 12 headers, 0 linesReliably Transmitting (NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK47b7ff4b;rportFrom: <sip:9048664201@4.78.224.161>;tag=as124705d0To: <sip:9048664201@4.78.224.161>Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1CSeq: 116 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 409 ConflictVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK47b7ff4b;rportFrom: <sip:9048664201@192.168.67.2>;tag=as124705d0To: <sip:9048664201@192.168.67.2>Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1CSeq: 116 REGISTERContent-Length: 0--- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161Destroying call '54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1'REGISTER 12 headers, 0 linesReliably Transmitting (NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4920bff9;rportFrom: <sip:9048664245@4.78.224.161>;tag=as60d8295aTo: <sip:9048664245@4.78.224.161>Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1CSeq: 112 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 409 ConflictVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK4920bff9;rportFrom: <sip:9048664245@192.168.67.2>;tag=as60d8295aTo: <sip:9048664245@192.168.67.2>Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1CSeq: 112 REGISTERContent-Length: 0--- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161Destroying call '69555ff937282717319589672fb5bae1@127.0.0.1'PP1*CLI><-- SIP read from 4.78.224.161:5060:INVITE sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69Session-Expires: 3600;Refresher=uacSupported: timerTo: <sip:s@4.78.224.161:5060>From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeCall-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dccContact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Type: application/sdpContent-Length: 144v=0o=dllsl3-ntsbc02 2147483647 2147483647 IN IP4 4.78.224.161s=sip callc=IN IP4 4.78.224.161t=0 0m=audio 30082 RTP/AVP 0a=ptime:20--- (16 headers 7 lines) ---Using INVITE request as basis request - 415225-3380410145-954166@dllsl3-ntsbc02.une3.comSending to 4.78.224.161 : 5060 (NAT)Found peer 'suntel-4245'Found RTP audio format 0Peer audio RTP is at port 4.78.224.161:30082Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)Looking for s in from-trunk (domain 4.78.224.161)list_route: hop: <sip:anonymous@4.78.224.161:5060>Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dcc;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeTo: <sip:s@4.78.224.161:5060>Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0--- -- Executing Set("SIP/9048664245-b68419b8", "FROM_DID=s") in new stack -- Executing Goto("SIP/9048664245-b68419b8", "ivr-3|s|1") in new stack -- Goto (ivr-3,s,1) -- Executing Set("SIP/9048664245-b68419b8", "LOOPCOUNT=0") in new stack -- Executing Set("SIP/9048664245-b68419b8", "__DIR-CONTEXT=zonemessages") in new stack -- Executing Answer("SIP/9048664245-b68419b8", "") in new stackWe're at 38.103.144.10 port 17826Adding codec 0x4 (ulaw) to SDPReliably Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dcc;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeTo: <sip:s@4.78.224.161:5060>;tag=as09c5756eCall-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Type: application/sdpContent-Length: 160v=0o=root 4971 4971 IN IP4 38.103.144.10s=sessionc=IN IP4 38.103.144.10t=0 0m=audio 17826 RTP/AVP 0a=rtpmap:0 PCMU/8000a=silenceSupp:off - - - ---- -- Executing Wait("SIP/9048664245-b68419b8", "1") in new stackPP1*CLI><-- SIP read from 4.78.224.161:5060:ACK sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69To: <sip:s@4.78.224.161:5060>;tag=as09c5756eFrom: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeCall-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 1 ACKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKe308536a98702bb4e9dfb62dd3668492Contact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Length: 0--- (13 headers 0 lines) --- -- Executing Set("SIP/9048664245-b68419b8", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing Set("SIP/9048664245-b68419b8", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/9048664245-b68419b8", "custom/SpanishEnterYourPinSunTel") in new stack -- Playing 'custom/SpanishEnterYourPinSunTel' (language 'en')PP1*CLI><-- SIP read from 4.78.224.161:5060:BYE sip:s@4.78.224.161:5060 SIP/2.0Max-Forwards: 69To: <sip:s@4.78.224.161:5060>;tag=as09c5756eFrom: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeCall-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 2 BYEVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKfc587f1ef7ee6f68eb31e8ddddffc9e0Contact: sip:anonymous@4.78.224.161:5060Allow-Events: line-seizeAllow-Events: dialogAllow-Events: referAllow-Events: message-summaryContent-Length: 0--- (13 headers 0 lines) ---Sending to 4.78.224.161 : 5060 (NAT)Transmitting (NAT) to 4.78.224.161:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKfc587f1ef7ee6f68eb31e8ddddffc9e0;received=4.78.224.161From: Anonymous <sip:anonymous@4.78.224.161;transport=udp>;tag=192.168.67.2+1+14dc52+9b54aeeTo: <sip:s@4.78.224.161:5060>;tag=as09c5756eCall-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.comCSeq: 2 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0--- == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/9048664245-b68419b8' -- Executing Hangup("SIP/9048664245-b68419b8", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9048664245-b68419b8'Destroying call '415225-3380410145-954166@dllsl3-ntsbc02.une3.com'PP1*CLI>