[root@MSNEMD-INTL191 ajpalmisano]# asterisk -rvvvvvvvvvvvvvvvvvvvv Asterisk certified/13.18-cert2, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk certified/13.18-cert2 currently running on MSNEMD-INTL191 (pid = 1321) MSNEMD-INTL191*CLI> sip set debug on SIP Debugging re-enabled MSNEMD-INTL191*CLI> sip reload Reloading SIP Reliably Transmitting (no NAT) to 162.223.83.235:5060: OPTIONS sip:bstnyc.granitevoip.com SIP/2.0 Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK3579acd9 Max-Forwards: 70 From: "asterisk" ;tag=as476b81b6 To: Contact: Call-ID: 4c94efd82667df273a1b850108685bd6@172.18.1.191:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX certified/13.18-cert2 Date: Mon, 29 Oct 2018 17:35:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:162.223.83.235:5060 ---> SIP/2.0 200 OK Call-ID: 4c94efd82667df273a1b850108685bd6@172.18.1.191:5060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as476b81b6 To: ;tag=sip+3+9f0200c3+254aee50 Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK3579acd9 Content-Length: 0 Supported: resource-priority, siprec, 100rel Server: DC-SIP/2.0 Organization: Metaswitch Networks Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/hook-flash, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml, application/vnd.telekom.service_indication+xml, application/calling-name-info <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '4c94efd82667df273a1b850108685bd6@172.18.1.191:5060' Method: OPTIONS [Oct 29 12:35:07] NOTICE[1429]: chan_sip.c:15724 sip_reregister: -- Re-registration for 6084715913@bstnyc.granitevoip.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 162.223.83.235:5060: REGISTER sip:bstnyc.granitevoip.com SIP/2.0 Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK6ae525f4 Max-Forwards: 70 From: ;tag=as4164df51 To: Call-ID: 0c5a1c1357b106036bb076775ac601bc@172.18.1.191 CSeq: 123 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX certified/13.18-cert2 Authorization: Digest username="6084715913", realm="bstnyc.granitevoip.com", algorithm=MD5, uri="sip:bstnyc.granitevoip.com", nonce="1a5d468454ac", response="b11dd347be07ed4b2e8347662c72abff", qop=auth, cnonce="554f944c", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:162.223.83.235:5060 ---> SIP/2.0 423 Interval Too Brief From: ;tag=as4164df51 To: ;tag=sip+3+b95800c4+a8534ad0 Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK6ae525f4 Server: SIP/2.0 Content-Length: 0 Call-ID: 0c5a1c1357b106036bb076775ac601bc@172.18.1.191 CSeq: 123 REGISTER Min-Expires: 3600 <-------------> --- (9 headers 0 lines) --- [Oct 29 12:35:07] WARNING[1429]: chan_sip.c:24453 handle_response_register: Got 423 Interval too brief for service 6084715913@bstnyc.granitevoip.com, minimum is 3600 seconds REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 162.223.83.235:5060: REGISTER sip:bstnyc.granitevoip.com SIP/2.0 Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK5e492ca5 Max-Forwards: 70 From: ;tag=as4164df51 To: Call-ID: 0c5a1c1357b106036bb076775ac601bc@172.18.1.191 CSeq: 124 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX certified/13.18-cert2 Authorization: Digest username="6084715913", realm="bstnyc.granitevoip.com", algorithm=MD5, uri="sip:bstnyc.granitevoip.com", nonce="1a5d468454ac", response="6ebf4187b297a8c30fe1a9ae11503fef", qop=auth, cnonce="0690f15f", nc=00000003 Expires: 3600 Contact: Content-Length: 0 --- Really destroying SIP dialog '0c5a1c1357b106036bb076775ac601bc@172.18.1.191' Method: REGISTER <--- SIP read from UDP:162.223.83.235:5060 ---> SIP/2.0 200 OK Call-ID: 0c5a1c1357b106036bb076775ac601bc@172.18.1.191 CSeq: 124 REGISTER From: ;tag=as4164df51 To: ;tag=sip+3+b95800c4+bf540c2e Via: SIP/2.0/UDP 172.18.1.191:5060;branch=z9hG4bK5e492ca5 Content-Length: 0 Expires: 3600 Contact: ;Expires=3600 Server: DC-SIP/2.0 Organization: Metaswitch Networks <-------------> --- (11 headers 0 lines) --- [Oct 29 12:35:07] NOTICE[1429]: chan_sip.c:24545 handle_response_register: Outbound Registration: Expiry for bstnyc.granitevoip.com is 3600 sec (Scheduling reregistration in 3585 s) Really destroying SIP dialog '0c5a1c1357b106036bb076775ac601bc@172.18.1.191' Method: REGISTER <--- SIP read from UDP:162.223.83.235:5060 ---> OPTIONS sip:6084715913@172.18.1.191:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+5602937e67509c7c9ac9742dd5d324461+sip+4+fddde095 From: ;tag=162.223.83.235+4+c0aeb9cd+4f5de115 Content-Length: 0 Supported: resource-priority, siprec, 100rel To: Contact: Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Max-Forwards: 69 Call-ID: 0gQAAC8WAAACBAAALxYAAOLLYzqRSuOQy++DQq6H1uyLS3FwTyb3oxqMGuPSok1s@162.223.83.235 CSeq: 995734591 OPTIONS Organization: Metaswitch Networks Accept: application/sdp, application/dtmf-relay <-------------> --- (13 headers 0 lines) --- Sending to 162.223.83.235:5060 (no NAT) Looking for 6084715913 in from-pstn (domain 172.18.1.191) <--- Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+5602937e67509c7c9ac9742dd5d324461+sip+4+fddde095;received=162.223.83.235 From: ;tag=162.223.83.235+4+c0aeb9cd+4f5de115 To: ;tag=as55f0ed32 Call-ID: 0gQAAC8WAAACBAAALxYAAOLLYzqRSuOQy++DQq6H1uyLS3FwTyb3oxqMGuPSok1s@162.223.83.235 CSeq: 995734591 OPTIONS Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0gQAAC8WAAACBAAALxYAAOLLYzqRSuOQy++DQq6H1uyLS3FwTyb3oxqMGuPSok1s@162.223.83.235' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '0gQAAC8WAAACBAAALxYAAHmdhqJQ7Ju7yTWx3OXeHF/a6LgMO6JYFWj0h5lT/SPj@162.223.83.235' Method: OPTIONS <--- SIP read from UDP:162.223.83.235:5060 ---> INVITE sip:6088880545@172.18.1.191:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+4f962253de6138e10193f39b02237bbd1+sip+2+fde0aaed From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: CSeq: 484877490 INVITE Expires: 180 Content-Length: 199 Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=2000" Supported: resource-priority,siprec, 100rel Contact: Content-Type: application/sdp Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 Organization: Metaswitch Networks Max-Forwards: 67 P-Asserted-Identity: "PALMISANO,SHELI" Accept: application/sdp, application/dtmf-relay v=0 o=- 64417454154211 64417454154211 IN IP4 162.223.83.240 s=- c=IN IP4 162.223.83.240 t=0 0 m=audio 56788 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=ptime:20 <-------------> --- (17 headers 9 lines) --- Sending to 162.223.83.235:5060 (no NAT) Sending to 162.223.83.235:5060 (no NAT) Using INVITE request as basis request - 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 Found peer 'SIP_Granite' for '5047028312' from 162.223.83.235:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7fc3f4008460 -- Strict RTP learning after remote address set to: 162.223.83.240:56788 Peer audio RTP is at port 162.223.83.240:56788 Looking for 6088880545 in from-pstn (domain 172.18.1.191) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+4f962253de6138e10193f39b02237bbd1+sip+2+fde0aaed;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877490 INVITE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [6088880545@from-pstn:1] Goto("SIP/SIP_Granite-00000001", "rightnumber,0,1") in new stack -- Goto (rightnumber,0,1) -- Executing [0@rightnumber:1] NoOp("SIP/SIP_Granite-00000001", ""Right number 6088880545"") in new stack -- Executing [0@rightnumber:2] Set("SIP/SIP_Granite-00000001", "intouchavailable=0") in new stack -- Executing [0@rightnumber:3] Ringing("SIP/SIP_Granite-00000001", "") in new stack <--- Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+4f962253de6138e10193f39b02237bbd1+sip+2+fde0aaed;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877490 INVITE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [0@rightnumber:4] Wait("SIP/SIP_Granite-00000001", "1") in new stack -- Executing [0@rightnumber:5] GotoIf("SIP/SIP_Granite-00000001", "1?livesystem,0,1") in new stack -- Goto (livesystem,0,1) -- Executing [0@livesystem:1] Answer("SIP/SIP_Granite-00000001", "") in new stack Audio is at 10598 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+4f962253de6138e10193f39b02237bbd1+sip+2+fde0aaed;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877490 INVITE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=root 517300783 517300783 IN IP4 172.18.1.191 s=Asterisk PBX certified/13.18-cert2 c=IN IP4 172.18.1.191 t=0 0 m=audio 10598 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:162.223.83.235:5060 ---> ACK sip:6088880545@172.18.1.191:5060 SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+adebfd31ce8fdb64155d872f5c8a7d901+sip+2+fde0ab6a Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 CSeq: 484877490 ACK Contact: Content-Length: 0 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Max-Forwards: 69 Organization: Metaswitch Networks <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:162.223.83.235:5060 ---> INVITE sip:6088880545@172.18.1.191:5060 SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+b83403b2c2f9919c4e3f38ad09ee03dd1+sip+2+fde0ab6e Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 CSeq: 484877491 INVITE Expires: 180 Content-Length: 175 Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=2000" Supported: resource-priority,siprec, 100rel Contact: Content-Type: application/sdp Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Max-Forwards: 69 Organization: Metaswitch Networks Accept: application/sdp, application/dtmf-relay v=0 o=- 64417454154211 64417454154212 IN IP4 162.223.83.240 s=- c=IN IP4 162.223.83.240 t=0 0 m=audio 56788 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 <-------------> --- (16 headers 8 lines) --- Sending to 162.223.83.235:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7fc3f4008460 -- Strict RTP learning after remote address set to: 162.223.83.240:56788 Peer audio RTP is at port 162.223.83.240:56788 <--- Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+b83403b2c2f9919c4e3f38ad09ee03dd1+sip+2+fde0ab6e;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877491 INVITE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 10598 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+b83403b2c2f9919c4e3f38ad09ee03dd1+sip+2+fde0ab6e;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877491 INVITE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=root 517300783 517300784 IN IP4 172.18.1.191 s=Asterisk PBX certified/13.18-cert2 c=IN IP4 172.18.1.191 t=0 0 m=audio 10598 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:162.223.83.235:5060 ---> ACK sip:6088880545@172.18.1.191:5060 SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+28a5388f1ff54bd714eac92b19e25f351+sip+2+fde0ab70 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 CSeq: 484877491 ACK Contact: Content-Length: 0 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Max-Forwards: 69 Organization: Metaswitch Networks <-------------> --- (11 headers 0 lines) --- -- Executing [0@livesystem:2] Goto("SIP/SIP_Granite-00000001", "nuance_landing,0,1") in new stack -- Goto (nuance_landing,0,1) -- Executing [0@nuance_landing:1] Wait("SIP/SIP_Granite-00000001", "8100") in new stack == Spawn extension (playbackfile, 0, 1) exited non-zero on 'SIP/SIP_Granite-00000001' -- Executing [0@playbackfile:1] ControlPlayback("SIP/SIP_Granite-00000001", ""/mnt/windows/PROMPTS/thankyouforcallingemdat",0,,,,,,o(0)") in new stack -- Playing '/mnt/windows/PROMPTS/thankyouforcallingemdat.slin' (language 'en') > 0x7fc3f4008460 -- Strict RTP switching to RTP target address 162.223.83.240:56788 as source -- Executing [0@playbackfile:2] Goto("SIP/SIP_Granite-00000001", "nuance_landing,0,1") in new stack -- Goto (nuance_landing,0,1) -- Executing [0@nuance_landing:1] Wait("SIP/SIP_Granite-00000001", "8100") in new stack == Spawn extension (playbackfile, 0, 1) exited non-zero on 'SIP/SIP_Granite-00000001' -- Executing [0@playbackfile:1] ControlPlayback("SIP/SIP_Granite-00000001", ""/mnt/windows/PROMPTS/enteryouruserid",0,,,,,,o(0)") in new stack -- Playing '/mnt/windows/PROMPTS/enteryouruserid.slin' (language 'en') Really destroying SIP dialog '0gQAAC8WAAACBAAALxYAAOLLYzqRSuOQy++DQq6H1uyLS3FwTyb3oxqMGuPSok1s@162.223.83.235' Method: OPTIONS > 0x7fc3f4008460 -- Strict RTP learning complete - Locking on source address 162.223.83.240:56788 -- Executing [0@playbackfile:2] Goto("SIP/SIP_Granite-00000001", "nuance_landing,0,1") in new stack -- Goto (nuance_landing,0,1) -- Executing [0@nuance_landing:1] Wait("SIP/SIP_Granite-00000001", "8100") in new stack <--- SIP read from UDP:162.223.83.235:5060 ---> BYE sip:6088880545@172.18.1.191:5060 SIP/2.0 Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+66b6f7e43b6a72d02a91b70d003ceb171+sip+2+fde0af5e Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 CSeq: 484877492 BYE Content-Length: 0 Supported: resource-priority, siprec, 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name Max-Forwards: 69 Organization: Metaswitch Networks <-------------> --- (11 headers 0 lines) --- Sending to 162.223.83.235:5060 (no NAT) Scheduling destruction of SIP dialog '0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 162.223.83.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 162.223.83.235:5060;branch=z9hG4bK+66b6f7e43b6a72d02a91b70d003ceb171+sip+2+fde0af5e;received=162.223.83.235 From: "PALMISANO,SHELI" ;tag=162.223.83.235+2+a423cc84+32d47bfd To: ;tag=as7b9549b1 Call-ID: 0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235 CSeq: 484877492 BYE Server: Asterisk PBX certified/13.18-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (nuance_landing, 0, 1) exited non-zero on 'SIP/SIP_Granite-00000001' Really destroying SIP dialog '0gQAAC8WAAACBAAALxYAANY5jbzKXkhxXYxL++wbTmgeaV4d4LJPDuc29xGhL9e3@162.223.83.235' Method: BYE -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected MSNEMD-INTL191*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups [root@MSNEMD-INTL191 ajpalmisano]#