<------------> -- Executing [4511@CommPanels:1] Dial("SIP/4510-0000000a", "SIP/4511, 20") in new stack == Using SIP RTP CoS mark 5 We think we can do text Audio is at 17800 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK20bf8de9;rport Max-Forwards: 70 From: "asterisk" ;tag=as639f6065 To: Contact: Call-ID: 135b8c1950a21f4164ca6b331b70f50b@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK20bf8de9;rport=5060 Contact: To: ;tag=ac48a430 From: "asterisk" ;tag=as639f6065 Call-ID: 135b8c1950a21f4164ca6b331b70f50b@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '135b8c1950a21f4164ca6b331b70f50b@10.1.1.36:5060' Method: OPTIONS Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec silk to SDP Adding codec silk to SDP Adding codec silk to SDP Adding codec silk to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.1.200.50:39873: INVITE sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5a1799be;rport Max-Forwards: 70 From: "4510" ;tag=as42bf828e To: Contact: Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 896 v=0 o=root 1063149757 1063149757 IN IP4 10.1.1.36 s=Asterisk PBX 13.18.2 c=IN IP4 10.1.1.36 t=0 0 m=audio 17800 RTP/AVP 0 8 3 4 111 112 5 120 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:120 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- -- Called SIP/4511 <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5a1799be;rport=5060 To: From: "4510" ;tag=as42bf828e Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK382de9c4;rport Max-Forwards: 70 From: "asterisk" ;tag=as3724930a To: Contact: Call-ID: 708b2ecd54681baf1519a096349ffac1@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK382de9c4 Call-ID: 708b2ecd54681baf1519a096349ffac1@10.1.1.36:5060 From: "asterisk" ;tag=as3724930a To: ;tag=z9hG4bK382de9c4 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '708b2ecd54681baf1519a096349ffac1@10.1.1.36:5060' Method: OPTIONS <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5a1799be;rport=5060 Contact: To: ;tag=c931443e From: "4510" ;tag=as42bf828e Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 102 INVITE User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/4511-0000000b is ringing <--- Transmitting (NAT) to 10.1.200.50:51872 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.200.50:51872;branch=z9hG4bKPj146fafdb79a84e5b850dba30a0979a35;received=10.1.200.50;rport=51872 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25408 INVITE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Length: 0 <------------> Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK66f16007;rport Max-Forwards: 70 From: "asterisk" ;tag=as47ba16b1 To: Contact: Call-ID: 1dab086831266ead74a457673d102626@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK66f16007;rport=5060 Contact: To: ;tag=c9584002 From: "asterisk" ;tag=as47ba16b1 Call-ID: 1dab086831266ead74a457673d102626@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '1dab086831266ead74a457673d102626@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK35e9bdba;rport Max-Forwards: 70 From: "asterisk" ;tag=as1479ea33 To: Contact: Call-ID: 6c0e56202e748f3f0dcda5472051750e@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK35e9bdba Call-ID: 6c0e56202e748f3f0dcda5472051750e@10.1.1.36:5060 From: "asterisk" ;tag=as1479ea33 To: ;tag=z9hG4bK35e9bdba CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '6c0e56202e748f3f0dcda5472051750e@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK2861a707;rport Max-Forwards: 70 From: "asterisk" ;tag=as0459a1ec To: Contact: Call-ID: 5f6c075c6c110e8434d3f00046810fe3@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK2861a707;rport=5060 Contact: To: ;tag=99799316 From: "asterisk" ;tag=as0459a1ec Call-ID: 5f6c075c6c110e8434d3f00046810fe3@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '5f6c075c6c110e8434d3f00046810fe3@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK558189ea;rport Max-Forwards: 70 From: "asterisk" ;tag=as0ed5ddaf To: Contact: Call-ID: 16bc8e9352ed97057b77374c4080c96b@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK558189ea Call-ID: 16bc8e9352ed97057b77374c4080c96b@10.1.1.36:5060 From: "asterisk" ;tag=as0ed5ddaf To: ;tag=z9hG4bK558189ea CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '16bc8e9352ed97057b77374c4080c96b@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5dd6e513;rport Max-Forwards: 70 From: "asterisk" ;tag=as5edc4931 To: Contact: Call-ID: 74c21f1715ef82070a205b120e871e7c@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5dd6e513;rport=5060 Contact: To: ;tag=3f77ee13 From: "asterisk" ;tag=as5edc4931 Call-ID: 74c21f1715ef82070a205b120e871e7c@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '74c21f1715ef82070a205b120e871e7c@10.1.1.36:5060' Method: OPTIONS <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5a1799be;rport=5060 Contact: To: ;tag=c931443e From: "4510" ;tag=as42bf828e Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 102 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 239 v=0 o=Z 0 2 IN IP4 78.189.8.164 s=Z c=IN IP4 78.189.8.164 t=0 0 m=audio 8000 RTP/AVP 0 3 110 8 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (11 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 101 Found audio description format speex for ID 110 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|speex|ilbc) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7f1e30008060 -- Strict RTP learning after remote address set to: 78.189.8.164:8000 Peer audio RTP is at port 78.189.8.164:8000 sip_route_dump: route/path hop: Transmitting (NAT) to 10.1.200.50:39873: ACK sip:4511@78.189.8.164:39873 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK4b6c8ee8;rport Max-Forwards: 70 From: "4510" ;tag=as42bf828e To: ;tag=c931443e Contact: Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.18.2 Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> PUBLISH sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---ea32735fa6907887 Max-Forwards: 70 Contact: To: From: ;tag=067e8d71 Call-ID: E9pgLhF8ls7y8SMRru2kVw.. CSeq: 1 PUBLISH Expires: 600 Content-Type: application/pidf+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence Allow-Events: presence, kpml, talk Content-Length: 263 open On the phone <-------------> --- (14 headers 3 lines) --- Sending to 10.1.200.50:39873 (NAT) <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---ea32735fa6907887;received=10.1.200.50;rport=39873 From: ;tag=067e8d71 To: ;tag=as0ab180c1 Call-ID: E9pgLhF8ls7y8SMRru2kVw.. CSeq: 1 PUBLISH Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---fc4b45d8f0b6c989 Max-Forwards: 70 Contact: To: From: ;tag=2112b520 Call-ID: MktVx9ZfH1y5AkCzM1PlKg.. CSeq: 1 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 10.1.200.50:39873 (NAT) Creating new subscription Sending to 10.1.200.50:39873 (NAT) sip_route_dump: route/path hop: Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---fc4b45d8f0b6c989;received=10.1.200.50;rport=39873 From: ;tag=2112b520 To: ;tag=as08ed2c9a Call-ID: MktVx9ZfH1y5AkCzM1PlKg.. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="241f6304" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MktVx9ZfH1y5AkCzM1PlKg..' in 6400 ms (Method: SUBSCRIBE) Really destroying SIP dialog 'E9pgLhF8ls7y8SMRru2kVw..' Method: PUBLISH -- SIP/4511-0000000b answered SIP/4510-0000000a <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---ec55351140702cc4 Max-Forwards: 70 Contact: To: From: ;tag=2112b520 Call-ID: MktVx9ZfH1y5AkCzM1PlKg.. CSeq: 2 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Authorization: Digest username="4511",realm="asterisk",nonce="241f6304",uri="sip:4511@10.1.1.36;transport=UDP",response="112ee8f01a067ebf50678ab4e7e6ab37",algorithm=MD5 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Creating new subscription Sending to 10.1.200.50:39873 (NAT) Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---ec55351140702cc4;received=10.1.200.50;rport=39873 From: ;tag=2112b520 To: ;tag=as08ed2c9a Call-ID: MktVx9ZfH1y5AkCzM1PlKg.. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog 'MktVx9ZfH1y5AkCzM1PlKg..' Method: SUBSCRIBE Audio is at 13466 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec opus to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.200.50:51872;branch=z9hG4bKPj146fafdb79a84e5b850dba30a0979a35;received=10.1.200.50;rport=51872 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25408 INVITE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Type: application/sdp Content-Length: 322 v=0 o=root 1298666104 1298666104 IN IP4 10.1.1.36 s=Asterisk PBX 13.18.2 c=IN IP4 10.1.1.36 t=0 0 m=audio 13466 RTP/AVP 0 8 123 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:60 a=sendrecv <------------> <--- SIP read from UDP:10.1.200.50:51872 ---> ACK sip:4511@10.1.1.36:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.200.50:51872;rport;branch=z9hG4bKPj039b386e8b9248229efd24510079e782 Max-Forwards: 70 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25408 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK69424d4d;rport Max-Forwards: 70 From: "asterisk" ;tag=as586b54f1 To: Contact: Call-ID: 2e13abff69e924c60369f2850dd0cf88@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> PUBLISH sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---a7a2f9d3c25d8454 Max-Forwards: 70 Contact: To: From: ;tag=560ceb3d Call-ID: oMcUiPhOvzCe-mINS5uHwQ.. CSeq: 1 PUBLISH Expires: 600 Content-Type: application/pidf+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence Allow-Events: presence, kpml, talk Content-Length: 263 open On the phone <-------------> --- (14 headers 3 lines) --- Sending to 10.1.200.50:39873 (NAT) <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---a7a2f9d3c25d8454;received=10.1.200.50;rport=39873 From: ;tag=560ceb3d To: ;tag=as5a91de34 Call-ID: oMcUiPhOvzCe-mINS5uHwQ.. CSeq: 1 PUBLISH Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---7a09638721c9f598 Max-Forwards: 70 Contact: To: From: ;tag=6a7a5a53 Call-ID: em_jDOEuVk1SFVGfzUMD6A.. CSeq: 1 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 10.1.200.50:39873 (NAT) Creating new subscription Sending to 10.1.200.50:39873 (NAT) sip_route_dump: route/path hop: Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---7a09638721c9f598;received=10.1.200.50;rport=39873 From: ;tag=6a7a5a53 To: ;tag=as7ef8b599 Call-ID: em_jDOEuVk1SFVGfzUMD6A.. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d118018" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'em_jDOEuVk1SFVGfzUMD6A..' in 6400 ms (Method: SUBSCRIBE) Really destroying SIP dialog 'oMcUiPhOvzCe-mINS5uHwQ..' Method: PUBLISH <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---5611abb422c26588 Max-Forwards: 70 Contact: To: From: ;tag=6a7a5a53 Call-ID: em_jDOEuVk1SFVGfzUMD6A.. CSeq: 2 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Authorization: Digest username="4511",realm="asterisk",nonce="7d118018",uri="sip:4511@10.1.1.36;transport=UDP",response="cc62d8e98b406539f5812833b4ef4abd",algorithm=MD5 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Creating new subscription Sending to 10.1.200.50:39873 (NAT) Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---5611abb422c26588;received=10.1.200.50;rport=39873 From: ;tag=6a7a5a53 To: ;tag=as7ef8b599 Call-ID: em_jDOEuVk1SFVGfzUMD6A.. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog 'em_jDOEuVk1SFVGfzUMD6A..' Method: SUBSCRIBE -- Channel SIP/4511-0000000b joined 'simple_bridge' basic-bridge <04a3898d-ed52-4b7f-9212-912d9fd56568> -- Channel SIP/4510-0000000a joined 'simple_bridge' basic-bridge <04a3898d-ed52-4b7f-9212-912d9fd56568> > Bridge 04a3898d-ed52-4b7f-9212-912d9fd56568: switching from simple_bridge technology to native_rtp > Locally RTP bridged 'SIP/4510-0000000a' and 'SIP/4511-0000000b' in stack > 0x7f1e38016510 -- Strict RTP switching to RTP target address 10.1.200.50:4010 as source > 0x7f1e38016510 -- Strict RTP learning complete - Locking on source address 10.1.200.50:4010 > 0x7f1e30008060 -- Strict RTP switching source address to 10.1.200.50:8000 <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK69424d4d Call-ID: 2e13abff69e924c60369f2850dd0cf88@10.1.1.36:5060 From: "asterisk" ;tag=as586b54f1 To: ;tag=z9hG4bK69424d4d CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.1.200.50:51872 ---> INVITE sip:4511@10.1.1.36:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.200.50:51872;rport;branch=z9hG4bKPjc85c9303663a46fab75fc68ed208fbcd Max-Forwards: 70 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Contact: "4510" Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25409 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 271 v=0 o=- 3720249282 3720249283 IN IP4 10.1.200.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 0 101 c=IN IP4 10.1.200.50 b=TIAS:64000 a=rtcp:4011 IN IP4 10.1.200.50 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 10.1.200.50:51872 (NAT) 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 - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), 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|) Peer audio RTP is at port 10.1.200.50:4010 <--- Transmitting (NAT) to 10.1.200.50:51872 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.200.50:51872;branch=z9hG4bKPjc85c9303663a46fab75fc68ed208fbcd;received=10.1.200.50;rport=51872 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25409 INVITE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 13466 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.200.50:51872;branch=z9hG4bKPjc85c9303663a46fab75fc68ed208fbcd;received=10.1.200.50;rport=51872 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25409 INVITE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 1298666104 1298666105 IN IP4 10.1.1.36 s=Asterisk PBX 13.18.2 c=IN IP4 10.1.1.36 t=0 0 m=audio 13466 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 <------------> Really destroying SIP dialog '2e13abff69e924c60369f2850dd0cf88@10.1.1.36:5060' Method: OPTIONS > Locally RTP bridged 'SIP/4510-0000000a' and 'SIP/4511-0000000b' in stack <--- SIP read from UDP:10.1.200.50:51872 ---> ACK sip:4511@10.1.1.36:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.200.50:51872;rport;branch=z9hG4bKPj6a20322785054d3e90103f3472137ca4 Max-Forwards: 70 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25409 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK75be709e;rport Max-Forwards: 70 From: "asterisk" ;tag=as17c78d17 To: Contact: Call-ID: 6f4db9cc6fb744931746dcb3770d8899@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK75be709e;rport=5060 Contact: To: ;tag=e576db44 From: "asterisk" ;tag=as17c78d17 Call-ID: 6f4db9cc6fb744931746dcb3770d8899@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '6f4db9cc6fb744931746dcb3770d8899@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK6f30f108;rport Max-Forwards: 70 From: "asterisk" ;tag=as15868822 To: Contact: Call-ID: 71a427a13950bd9e5559c5b21af1efd4@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK6f30f108 Call-ID: 71a427a13950bd9e5559c5b21af1efd4@10.1.1.36:5060 From: "asterisk" ;tag=as15868822 To: ;tag=z9hG4bK6f30f108 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '71a427a13950bd9e5559c5b21af1efd4@10.1.1.36:5060' Method: OPTIONS > 0x7f1e30008060 -- Strict RTP learning complete - Locking on source address 10.1.200.50:8000 Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK79aac2d8;rport Max-Forwards: 70 From: "asterisk" ;tag=as3e22762a To: Contact: Call-ID: 0ed57d007f803976423562a04711ca04@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK79aac2d8;rport=5060 Contact: To: ;tag=d531bd2f From: "asterisk" ;tag=as3e22762a Call-ID: 0ed57d007f803976423562a04711ca04@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '0ed57d007f803976423562a04711ca04@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK4221a7a5;rport Max-Forwards: 70 From: "asterisk" ;tag=as1bfd2b13 To: Contact: Call-ID: 05aa743b4abe9f326046f6f20900548c@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK4221a7a5 Call-ID: 05aa743b4abe9f326046f6f20900548c@10.1.1.36:5060 From: "asterisk" ;tag=as1bfd2b13 To: ;tag=z9hG4bK4221a7a5 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '05aa743b4abe9f326046f6f20900548c@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK21be9a07;rport Max-Forwards: 70 From: "asterisk" ;tag=as455a6ef8 To: Contact: Call-ID: 5792251b08e090e811bbaf602c3c476e@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK21be9a07;rport=5060 Contact: To: ;tag=243a121c From: "asterisk" ;tag=as455a6ef8 Call-ID: 5792251b08e090e811bbaf602c3c476e@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '5792251b08e090e811bbaf602c3c476e@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK21138f56;rport Max-Forwards: 70 From: "asterisk" ;tag=as2f859f4c To: Contact: Call-ID: 4c06e63d3de812491d8d753e1567b425@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK21138f56 Call-ID: 4c06e63d3de812491d8d753e1567b425@10.1.1.36:5060 From: "asterisk" ;tag=as2f859f4c To: ;tag=z9hG4bK21138f56 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '4c06e63d3de812491d8d753e1567b425@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK515a1a35;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a08eb81 To: Contact: Call-ID: 0ef245306fb2fde1572fb94660ed54ab@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK515a1a35;rport=5060 Contact: To: ;tag=bd513f0e From: "asterisk" ;tag=as0a08eb81 Call-ID: 0ef245306fb2fde1572fb94660ed54ab@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '0ef245306fb2fde1572fb94660ed54ab@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK16a380c2;rport Max-Forwards: 70 From: "asterisk" ;tag=as58059924 To: Contact: Call-ID: 385780154b187d0144b3129c13275d77@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK16a380c2 Call-ID: 385780154b187d0144b3129c13275d77@10.1.1.36:5060 From: "asterisk" ;tag=as58059924 To: ;tag=z9hG4bK16a380c2 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '385780154b187d0144b3129c13275d77@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK2e494f11;rport Max-Forwards: 70 From: "asterisk" ;tag=as7022e46f To: Contact: Call-ID: 6581965f69077c9d0d5570333461d8b2@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK2e494f11;rport=5060 Contact: To: ;tag=75640959 From: "asterisk" ;tag=as7022e46f Call-ID: 6581965f69077c9d0d5570333461d8b2@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '6581965f69077c9d0d5570333461d8b2@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK37a658d0;rport Max-Forwards: 70 From: "asterisk" ;tag=as11f2fb8e To: Contact: Call-ID: 1aaf78916f9b939d105b2af84a312ad3@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK37a658d0 Call-ID: 1aaf78916f9b939d105b2af84a312ad3@10.1.1.36:5060 From: "asterisk" ;tag=as11f2fb8e To: ;tag=z9hG4bK37a658d0 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '1aaf78916f9b939d105b2af84a312ad3@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK7f765078;rport Max-Forwards: 70 From: "asterisk" ;tag=as6a884c10 To: Contact: Call-ID: 19ee237133f3dbfd183bed71205b2869@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK7f765078;rport=5060 Contact: To: ;tag=6b46a619 From: "asterisk" ;tag=as6a884c10 Call-ID: 19ee237133f3dbfd183bed71205b2869@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '19ee237133f3dbfd183bed71205b2869@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK3978b106;rport Max-Forwards: 70 From: "asterisk" ;tag=as060f1aa9 To: Contact: Call-ID: 2ef8451a2eef021521c69edb2999bb6b@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK3978b106 Call-ID: 2ef8451a2eef021521c69edb2999bb6b@10.1.1.36:5060 From: "asterisk" ;tag=as060f1aa9 To: ;tag=z9hG4bK3978b106 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '2ef8451a2eef021521c69edb2999bb6b@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK7668965b;rport Max-Forwards: 70 From: "asterisk" ;tag=as4c5bde4b To: Contact: Call-ID: 3a3af4fd74700f1a308403166c98d057@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK7668965b;rport=5060 Contact: To: ;tag=3d1f795b From: "asterisk" ;tag=as4c5bde4b Call-ID: 3a3af4fd74700f1a308403166c98d057@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '3a3af4fd74700f1a308403166c98d057@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK08c36fa3;rport Max-Forwards: 70 From: "asterisk" ;tag=as1cbdce5b To: Contact: Call-ID: 7f9061522cac12df4628e96a07c1dadb@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK08c36fa3 Call-ID: 7f9061522cac12df4628e96a07c1dadb@10.1.1.36:5060 From: "asterisk" ;tag=as1cbdce5b To: ;tag=z9hG4bK08c36fa3 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '7f9061522cac12df4628e96a07c1dadb@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK378fc14e;rport Max-Forwards: 70 From: "asterisk" ;tag=as40273afa To: Contact: Call-ID: 0abdfd38538f02115c8c85c466dc2077@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK378fc14e;rport=5060 Contact: To: ;tag=da73fb72 From: "asterisk" ;tag=as40273afa Call-ID: 0abdfd38538f02115c8c85c466dc2077@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '0abdfd38538f02115c8c85c466dc2077@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK7e75ad7b;rport Max-Forwards: 70 From: "asterisk" ;tag=as73193268 To: Contact: Call-ID: 399d3ffb220ccc05276c955749baec8d@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK7e75ad7b Call-ID: 399d3ffb220ccc05276c955749baec8d@10.1.1.36:5060 From: "asterisk" ;tag=as73193268 To: ;tag=z9hG4bK7e75ad7b CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '399d3ffb220ccc05276c955749baec8d@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK6ad6d9e7;rport Max-Forwards: 70 From: "asterisk" ;tag=as36075aa7 To: Contact: Call-ID: 2866d0c80622e9f77eba6b054c90f75b@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK6ad6d9e7;rport=5060 Contact: To: ;tag=fc5b676b From: "asterisk" ;tag=as36075aa7 Call-ID: 2866d0c80622e9f77eba6b054c90f75b@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '2866d0c80622e9f77eba6b054c90f75b@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK6ea1f580;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ca7cc0a To: Contact: Call-ID: 568e61017169c6af689be6584951b66c@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK6ea1f580 Call-ID: 568e61017169c6af689be6584951b66c@10.1.1.36:5060 From: "asterisk" ;tag=as6ca7cc0a To: ;tag=z9hG4bK6ea1f580 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '568e61017169c6af689be6584951b66c@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK28e05b74;rport Max-Forwards: 70 From: "asterisk" ;tag=as349dc162 To: Contact: Call-ID: 7843bfd0695a5b0c7b7d065168ab7a7a@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK28e05b74;rport=5060 Contact: To: ;tag=856d1a39 From: "asterisk" ;tag=as349dc162 Call-ID: 7843bfd0695a5b0c7b7d065168ab7a7a@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '7843bfd0695a5b0c7b7d065168ab7a7a@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK76d02584;rport Max-Forwards: 70 From: "asterisk" ;tag=as3791acc1 To: Contact: Call-ID: 6fad4d8c1da973d400bc53776c49a9c0@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK76d02584 Call-ID: 6fad4d8c1da973d400bc53776c49a9c0@10.1.1.36:5060 From: "asterisk" ;tag=as3791acc1 To: ;tag=z9hG4bK76d02584 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '6fad4d8c1da973d400bc53776c49a9c0@10.1.1.36:5060' Method: OPTIONS <--- SIP read from UDP:10.1.200.50:51872 ---> BYE sip:4511@10.1.1.36:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.200.50:51872;rport;branch=z9hG4bKPjdccb3d426e1045d88a4dd40bf3def64c Max-Forwards: 70 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25410 BYE User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.1.200.50:51872 (NAT) Scheduling destruction of SIP dialog 'ac5fa8a7a03a417c80f1ab3be13458bb' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.200.50:51872;branch=z9hG4bKPjdccb3d426e1045d88a4dd40bf3def64c;received=10.1.200.50;rport=51872 From: "4510" ;tag=d5356db53b024142858e904a91f55c60 To: ;tag=as53d9d0d8 Call-ID: ac5fa8a7a03a417c80f1ab3be13458bb CSeq: 25410 BYE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> -- Channel SIP/4510-0000000a left 'native_rtp' basic-bridge <04a3898d-ed52-4b7f-9212-912d9fd56568> == Spawn extension (CommPanels, 4511, 1) exited non-zero on 'SIP/4510-0000000a' -- Channel SIP/4511-0000000b left 'native_rtp' basic-bridge <04a3898d-ed52-4b7f-9212-912d9fd56568> Scheduling destruction of SIP dialog '501574451a2e7c4867e9760a099b608e@10.1.1.36:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.1.200.50:39873: BYE sip:4511@78.189.8.164:39873 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK0cf6cac5;rport Max-Forwards: 70 From: "4510" ;tag=as42bf828e To: ;tag=c931443e Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 13.18.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK0cf6cac5;rport=5060 Contact: To: ;tag=c931443e From: "4510" ;tag=as42bf828e Call-ID: 501574451a2e7c4867e9760a099b608e@10.1.1.36:5060 CSeq: 103 BYE User-Agent: Z 3.15.40006 rv2.8.20 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '501574451a2e7c4867e9760a099b608e@10.1.1.36:5060' Method: INVITE <--- SIP read from UDP:10.1.200.50:39873 ---> PUBLISH sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---fbba830e7ca333b1 Max-Forwards: 70 Contact: To: From: ;tag=fb18ff7d Call-ID: hJtWlXpIgYdOuueRkCjmWQ.. CSeq: 1 PUBLISH Expires: 600 Content-Type: application/pidf+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence Allow-Events: presence, kpml, talk Content-Length: 257 open Online <-------------> --- (14 headers 3 lines) --- Sending to 10.1.200.50:39873 (NAT) <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---fbba830e7ca333b1;received=10.1.200.50;rport=39873 From: ;tag=fb18ff7d To: ;tag=as7f1bee7d Call-ID: hJtWlXpIgYdOuueRkCjmWQ.. CSeq: 1 PUBLISH Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---a54a64d5f0d924a3 Max-Forwards: 70 Contact: To: From: ;tag=9f18c63a Call-ID: t2ZwXScoxeoBfMlcfKk1tA.. CSeq: 1 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 10.1.200.50:39873 (NAT) Creating new subscription Sending to 10.1.200.50:39873 (NAT) sip_route_dump: route/path hop: Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---a54a64d5f0d924a3;received=10.1.200.50;rport=39873 From: ;tag=9f18c63a To: ;tag=as78132c0c Call-ID: t2ZwXScoxeoBfMlcfKk1tA.. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c609c37" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 't2ZwXScoxeoBfMlcfKk1tA..' in 6400 ms (Method: SUBSCRIBE) Really destroying SIP dialog 'hJtWlXpIgYdOuueRkCjmWQ..' Method: PUBLISH <--- SIP read from UDP:10.1.200.50:39873 ---> SUBSCRIBE sip:4511@10.1.1.36;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---3806b476be4a6707 Max-Forwards: 70 Contact: To: From: ;tag=9f18c63a Call-ID: t2ZwXScoxeoBfMlcfKk1tA.. CSeq: 2 SUBSCRIBE Expires: 600 Accept: application/watcherinfo+xml User-Agent: Z 3.15.40006 rv2.8.20 Authorization: Digest username="4511",realm="asterisk",nonce="5c609c37",uri="sip:4511@10.1.1.36;transport=UDP",response="6b28545a49eb0e272d54e70e442a26aa",algorithm=MD5 Event: presence.winfo Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Creating new subscription Sending to 10.1.200.50:39873 (NAT) Found peer '4511' for '4511' from 10.1.200.50:39873 <--- Transmitting (NAT) to 10.1.200.50:39873 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 78.189.8.164:39873;branch=z9hG4bK-524287-1---3806b476be4a6707;received=10.1.200.50;rport=39873 From: ;tag=9f18c63a To: ;tag=as78132c0c Call-ID: t2ZwXScoxeoBfMlcfKk1tA.. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 13.18.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog 't2ZwXScoxeoBfMlcfKk1tA..' Method: SUBSCRIBE Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK65cfd692;rport Max-Forwards: 70 From: "asterisk" ;tag=as4d8cc76a To: Contact: Call-ID: 2c4c376b6ee79cf440740dd334f58bae@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK65cfd692;rport=5060 Contact: To: ;tag=41196217 From: "asterisk" ;tag=as4d8cc76a Call-ID: 2c4c376b6ee79cf440740dd334f58bae@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '2c4c376b6ee79cf440740dd334f58bae@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK21f4a0ac;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c5d899f To: Contact: Call-ID: 2033225d512b4b5c0c31eeb945ce9978@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK21f4a0ac Call-ID: 2033225d512b4b5c0c31eeb945ce9978@10.1.1.36:5060 From: "asterisk" ;tag=as3c5d899f To: ;tag=z9hG4bK21f4a0ac CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '2033225d512b4b5c0c31eeb945ce9978@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5527435a;rport Max-Forwards: 70 From: "asterisk" ;tag=as0b7f3300 To: Contact: Call-ID: 4a8164c9249328db24a01d896abd0087@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK5527435a;rport=5060 Contact: To: ;tag=e94b9756 From: "asterisk" ;tag=as0b7f3300 Call-ID: 4a8164c9249328db24a01d896abd0087@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '4a8164c9249328db24a01d896abd0087@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK219f1fc6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0b27b3ac To: Contact: Call-ID: 5dd3ae7b731ba4a57406e98c5c1fae01@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK219f1fc6 Call-ID: 5dd3ae7b731ba4a57406e98c5c1fae01@10.1.1.36:5060 From: "asterisk" ;tag=as0b27b3ac To: ;tag=z9hG4bK219f1fc6 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '5dd3ae7b731ba4a57406e98c5c1fae01@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK15a11d83;rport Max-Forwards: 70 From: "asterisk" ;tag=as13709857 To: Contact: Call-ID: 3960d2bd2311b3b05c855b2460ea5f57@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK15a11d83;rport=5060 Contact: To: ;tag=712ca233 From: "asterisk" ;tag=as13709857 Call-ID: 3960d2bd2311b3b05c855b2460ea5f57@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '3960d2bd2311b3b05c855b2460ea5f57@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:51872: OPTIONS sip:4510@10.1.200.50:51872;ob SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK0bfdb55f;rport Max-Forwards: 70 From: "asterisk" ;tag=as59f87130 To: Contact: Call-ID: 0d3ea49675371d384605054553ed6b76@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:51872 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;received=10.1.1.36;branch=z9hG4bK0bfdb55f Call-ID: 0d3ea49675371d384605054553ed6b76@10.1.1.36:5060 From: "asterisk" ;tag=as59f87130 To: ;tag=z9hG4bK0bfdb55f CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.14.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '0d3ea49675371d384605054553ed6b76@10.1.1.36:5060' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.200.50:39873: OPTIONS sip:4511@10.1.200.50:39873;rinstance=0a7de5aec2fc1401;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK444d1363;rport Max-Forwards: 70 From: "asterisk" ;tag=as41dd617b To: Contact: Call-ID: 66232a496ca958694437112018cf46ba@10.1.1.36:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.18.2 Date: Tue, 21 Nov 2017 07:33:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.1.200.50:39873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.36:5060;branch=z9hG4bK444d1363;rport=5060 Contact: To: ;tag=95755f1b From: "asterisk" ;tag=as41dd617b Call-ID: 66232a496ca958694437112018cf46ba@10.1.1.36:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '66232a496ca958694437112018cf46ba@10.1.1.36:5060' Method: OPTIONS asterisk-test-server*CLI> sip set debug off SIP Debugging Disabled