Really destroying SIP dialog '5d41540d09b5fcd90ac46a9279ad8328@10.2.0.20:5060' Method: OPTIONS dclvccast*CLI> dclvccast*CLI> dclvccast*CLI> Really destroying SIP dialog '76bc3d9520020a592613db044d0fe3fd@10.2.0.20:5060' Method: OPTIONS Really destroying SIP dialog '1a421a915abcf87d6d34f16d082d59a3@10.2.0.20:5060' Method: OPTIONS Really destroying SIP dialog '685a82dd78171fe2277282ff718f33a7@62.122.184.91' Method: OPTIONS <--- SIP read from UDP:10.0.11.68:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.91:5060 ---> <-------------> Really destroying SIP dialog '41a8c50002c26f8d40050d1c77bac7a7@10.222.0.20:5060' Method: OPTIONS <--- SIP read from UDP:10.0.11.80:5060 ---> <-------------> <--- SIP read from UDP:10.0.7.91:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.74:5060 ---> <-------------> <--- SIP read from UDP:62.105.136.42:5060 ---> OPTIONS sip:195.13.217.88 SIP/2.0 Via: SIP/2.0/UDP 62.105.136.42:5060;branch=z9hG4bK259e6d31;rport Max-Forwards: 70 From: "ipcom" ;tag=as6580d1f0 To: Contact: Call-ID: 6853b76f5cb7486a5d49bfd972827311@62.105.136.42:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg-1~bpo60+1 Date: Wed, 13 Apr 2016 05:49:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 62.105.136.42:5060 (no NAT) Looking for s in incoming (domain 195.13.217.88) <--- Transmitting (no NAT) to 62.105.136.42:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 62.105.136.42:5060;branch=z9hG4bK259e6d31;received=62.105.136.42;rport=5060 From: "ipcom" ;tag=as6580d1f0 To: ;tag=as5f62c971 Call-ID: 6853b76f5cb7486a5d49bfd972827311@62.105.136.42:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6853b76f5cb7486a5d49bfd972827311@62.105.136.42:5060' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '587b599e44e2f1511268ed355c0dc71a@10.111.0.7:5060' Method: OPTIONS <--- SIP read from UDP:10.0.11.82:5060 ---> <-------------> Really destroying SIP dialog '5332789229644-5182162026088@10.1.0.22' Method: REGISTER Really destroying SIP dialog '528f676f1939710c09aaa1bd7bb13aea@10.222.0.7:5060' Method: OPTIONS Really destroying SIP dialog '371718ed41e970bd55f4e5dd3d9dfd26@77.41.172.198:5060' Method: OPTIONS Really destroying SIP dialog '46131980@10.111.0.7' Method: OPTIONS <--- SIP read from UDP:10.0.11.86:5060 ---> <-------------> [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:15580 sip_reregister: -- Re-registration for 4956611065@vats437.vats.cifra1.ru REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 91.143.36.250:5060: REGISTER sip:vats437.vats.cifra1.ru SIP/2.0 Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK103593c0 Max-Forwards: 70 From: ;tag=as0f16c86c To: Call-ID: 66f45e6c359c6abd63f0c06a5872b1e4@10.111.0.7 CSeq: 104 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.8.0 Authorization: Digest username="4956611065", realm="vats437.vats.cifra1.ru", algorithm=MD5, uri="sip:vats437.vats.cifra1.ru", nonce="Vw3eXFcN3TApDoaUjK1UwRyrFDjA5Pg/", response="ec7f6e2850b08fe6b1537bd9a6d38738" Expires: 1800 Contact: Content-Length: 0 --- <--- SIP read from UDP:91.143.36.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK103593c0;rport=5060 From: ;tag=as0f16c86c To: ;tag=553b5a67c6d93e4f93f9dd674b4083de.8ca4 Call-ID: 66f45e6c359c6abd63f0c06a5872b1e4@10.111.0.7 CSeq: 104 REGISTER Contact: ;expires=180 Server: Sip-Server Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:24356 handle_response_register: Outbound Registration: Expiry for vats437.vats.cifra1.ru is 180 sec (Scheduling reregistration in 165 s) Really destroying SIP dialog '66f45e6c359c6abd63f0c06a5872b1e4@10.111.0.7' Method: REGISTER [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:15580 sip_reregister: -- Re-registration for 7810051@195.90.150.205 REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to 195.90.150.205:5060: REGISTER sip:195.90.150.205 SIP/2.0 Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK37069b4e Max-Forwards: 70 From: ;tag=as4473f612 To: Call-ID: 242599d5644902211fc245f24c57dcef@10.111.0.7 CSeq: 103 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.8.0 Expires: 1800 Contact: Content-Length: 0 --- <--- SIP read from UDP:195.90.150.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK37069b4e From: ;tag=as4473f612 To: ;tag=aprqtkboj91-acmokq30000e6 Call-ID: 242599d5644902211fc245f24c57dcef@10.111.0.7 CSeq: 103 REGISTER Contact: ;expires=180 Expires: 180 <-------------> --- (8 headers 0 lines) --- [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:24356 handle_response_register: Outbound Registration: Expiry for 195.90.150.205 is 180 sec (Scheduling reregistration in 165 s) Really destroying SIP dialog '242599d5644902211fc245f24c57dcef@10.111.0.7' Method: REGISTER [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:15580 sip_reregister: -- Re-registration for 4957307567@vats437.vats.cifra1.ru REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 91.143.36.250:5060: REGISTER sip:vats437.vats.cifra1.ru SIP/2.0 Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK6e8c8f4b Max-Forwards: 70 From: ;tag=as7ba192ac To: Call-ID: 2cedac9208e0869a6a65de0f3c228b35@10.111.0.7 CSeq: 104 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.8.0 Authorization: Digest username="4957307567", realm="vats437.vats.cifra1.ru", algorithm=MD5, uri="sip:vats437.vats.cifra1.ru", nonce="Vw3eXVcN3TEwWlVjHOR8Xjk7hfraUR+/", response="0550fb17ff0e0615f7965d16d552b75a" Expires: 1800 Contact: Content-Length: 0 --- <--- SIP read from UDP:91.143.36.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK6e8c8f4b;rport=5060 From: ;tag=as7ba192ac To: ;tag=553b5a67c6d93e4f93f9dd674b4083de.8fe7 Call-ID: 2cedac9208e0869a6a65de0f3c228b35@10.111.0.7 CSeq: 104 REGISTER Contact: ;expires=180 Server: Sip-Server Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Apr 13 08:49:10] NOTICE[29083]: chan_sip.c:24356 handle_response_register: Outbound Registration: Expiry for vats437.vats.cifra1.ru is 180 sec (Scheduling reregistration in 165 s) Really destroying SIP dialog '2cedac9208e0869a6a65de0f3c228b35@10.111.0.7' Method: REGISTER Really destroying SIP dialog 'NDAwNDA1YTU2ZWY5YmNkMjJjMTY1Y2I4ZGEzNTExOGI.' Method: REGISTER Really destroying SIP dialog '1dd1d6ad49378fbd21d87c205270e267@62.122.184.91' Method: OPTIONS == Manager 'web' logged off from 10.222.0.8 == Manager 'web' logged on from 10.222.0.8 <--- SIP read from UDP:10.0.8.63:56964 ---> <-------------> <--- SIP read from UDP:10.0.11.68:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.91:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.80:5060 ---> <-------------> <--- SIP read from UDP:10.0.7.91:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.74:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.82:5060 ---> <-------------> <--- SIP read from UDP:10.0.11.86:5060 ---> <-------------> Reliably Transmitting (no NAT) to 10.0.11.83:5060: OPTIONS sip:2828@10.0.11.83:5060 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK428fb498 Max-Forwards: 70 From: "asterisk" ;tag=as63b97542 To: Contact: Call-ID: 72aee2dc45530a0c44c3c98e11891c2e@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.0.11.83:5060 ---> SIP/2.0 200 OK To: ;tag=bb331d317dfcdeaei0 From: "asterisk" ;tag=as63b97542 Call-ID: 72aee2dc45530a0c44c3c98e11891c2e@10.111.0.7:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK428fb498 Server: Cisco/SPA303-7.5.2 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '72aee2dc45530a0c44c3c98e11891c2e@10.111.0.7:5060' Method: OPTIONS <--- SIP read from UDP:10.2.0.20:5060 ---> OPTIONS sip:222222@10.111.0.7:5060 SIP/2.0 Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK31623a13 Max-Forwards: 70 From: "asterisk" ;tag=as45ac1fab To: Contact: Call-ID: 4f31a967307529785d376a82546de357@10.2.0.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Date: Wed, 13 Apr 2016 05:49:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.2.0.20:5060 (no NAT) Looking for 222222 in incoming (domain 10.111.0.7) <--- Transmitting (no NAT) to 10.2.0.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK31623a13;received=10.2.0.20 From: "asterisk" ;tag=as45ac1fab To: ;tag=as120bb3ee Call-ID: 4f31a967307529785d376a82546de357@10.2.0.20:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.8.0 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 '4f31a967307529785d376a82546de357@10.2.0.20:5060' in 32000 ms (Method: OPTIONS) Reliably Transmitting (no NAT) to 10.0.8.74:5060: OPTIONS sip:2829@10.0.8.74:5060 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK069933bb Max-Forwards: 70 From: "asterisk" ;tag=as71183ded To: Contact: Call-ID: 2df2e1d36853bed45a0d08603bd0207e@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 10.4.12.20:5060: OPTIONS sip:2825@10.4.12.20:5060 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK303c7a6a Max-Forwards: 70 From: "asterisk" ;tag=as411fd8e5 To: Contact: Call-ID: 6a1e220b785789ee3221f1e76258a29c@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.2.0.20:5060 ---> OPTIONS sip:111111@10.111.0.7:5060 SIP/2.0 Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK7c9da613 Max-Forwards: 70 From: "asterisk" ;tag=as2353912f To: Contact: Call-ID: 123e747a1599310270af2b9f0c0a1c56@10.2.0.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Date: Wed, 13 Apr 2016 05:49:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.2.0.20:5060 (no NAT) Looking for 111111 in incoming (domain 10.111.0.7) <--- Transmitting (no NAT) to 10.2.0.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK7c9da613;received=10.2.0.20 From: "asterisk" ;tag=as2353912f To: ;tag=as37c7026b Call-ID: 123e747a1599310270af2b9f0c0a1c56@10.2.0.20:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.8.0 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 '123e747a1599310270af2b9f0c0a1c56@10.2.0.20:5060' in 32000 ms (Method: OPTIONS) <--- SIP read from UDP:10.0.8.74:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK069933bb Call-ID: 2df2e1d36853bed45a0d08603bd0207e@10.111.0.7:5060 From: "asterisk" ;tag=as71183ded To: ;tag=2348873863 CSeq: 102 OPTIONS Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER Contact: Server: Panasonic_KX-UT123RU/01.278 (080023a4797e) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '2df2e1d36853bed45a0d08603bd0207e@10.111.0.7:5060' Method: OPTIONS <--- SIP read from UDP:10.4.12.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK303c7a6a Call-ID: 6a1e220b785789ee3221f1e76258a29c@10.111.0.7:5060 From: "asterisk" ;tag=as411fd8e5 To: ;tag=4111449611 CSeq: 102 OPTIONS Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER Contact: Server: Panasonic_KX-UT123RU/01.278 (080023a468d3) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '6a1e220b785789ee3221f1e76258a29c@10.111.0.7:5060' Method: OPTIONS <--- SIP read from UDP:10.2.0.20:5060 ---> OPTIONS sip:333333@10.111.0.7:5060 SIP/2.0 Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK1d3e19dd Max-Forwards: 70 From: "asterisk" ;tag=as6d53ed4f To: Contact: Call-ID: 3ca5f1211b29e9234cf00ca04dc9c1ab@10.2.0.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Date: Wed, 13 Apr 2016 05:49:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.2.0.20:5060 (no NAT) Looking for 333333 in incoming (domain 10.111.0.7) <--- Transmitting (no NAT) to 10.2.0.20:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.2.0.20:5060;branch=z9hG4bK1d3e19dd;received=10.2.0.20 From: "asterisk" ;tag=as6d53ed4f To: ;tag=as5a821b5d Call-ID: 3ca5f1211b29e9234cf00ca04dc9c1ab@10.2.0.20:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3ca5f1211b29e9234cf00ca04dc9c1ab@10.2.0.20:5060' in 32000 ms (Method: OPTIONS) Reliably Transmitting (no NAT) to 10.2.0.20:5060: OPTIONS sip:10.2.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK2cef4d52 Max-Forwards: 70 From: "asterisk" ;tag=as5542497f To: Contact: Call-ID: 1a5edfea63589d394ea678637927aa2a@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.2.0.20:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK2cef4d52;received=10.111.0.7 From: "asterisk" ;tag=as5542497f To: ;tag=as26d96df2 Call-ID: 1a5edfea63589d394ea678637927aa2a@10.111.0.7:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '1a5edfea63589d394ea678637927aa2a@10.111.0.7:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 77.94.2.34:5060: OPTIONS sip:77.94.2.34 SIP/2.0 Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK0134ff9e Max-Forwards: 70 From: "asterisk" ;tag=as33d0cb6e To: Contact: Call-ID: 439882b74a6606bc5f059fc75d19b325@195.13.217.88:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:77.94.2.34:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK0134ff9e From: "asterisk" ;tag=as33d0cb6e To: ;tag=3157747834-3859888897-4082893236-1683941628 Call-ID: 439882b74a6606bc5f059fc75d19b325@195.13.217.88:5060 CSeq: 102 OPTIONS Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE Accept: application/dtmf-relay Accept: application/ISUP Accept: application/sdp Supported: 100rel Server: MERA MVTS3G v.4.4.0-24 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '439882b74a6606bc5f059fc75d19b325@195.13.217.88:5060' Method: OPTIONS <--- SIP read from UDP:10.0.11.91:5060 ---> BYE sip:asterisk@10.111.0.7:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.11.91:5060;rport;branch=z9hG4bKPjc56e9ca1d1b046c5978c0b57f873e190 Max-Forwards: 70 From: ;tag=a0ce31b00adf4cac9b26ad388ba761e2 To: "DOMOVOY89137529220" ;tag=as257a343a Call-ID: 77d0b55809059c0b5f4edfe07e86b593@10.111.0.7:5060 CSeq: 10223 BYE User-Agent: Domovoy SIP Phone v0.9.0-release/win32 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.0.11.91:5060 (no NAT) Scheduling destruction of SIP dialog '77d0b55809059c0b5f4edfe07e86b593@10.111.0.7:5060' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 10.0.11.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.11.91:5060;branch=z9hG4bKPjc56e9ca1d1b046c5978c0b57f873e190;received=10.0.11.91;rport=5060 From: ;tag=a0ce31b00adf4cac9b26ad388ba761e2 To: "DOMOVOY89137529220" ;tag=as257a343a Call-ID: 77d0b55809059c0b5f4edfe07e86b593@10.111.0.7:5060 CSeq: 10223 BYE Server: Asterisk PBX 13.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/2815-00000005 left 'simple_bridge' basic-bridge -- Channel SIP/sib-83832331290-00000006 left 'simple_bridge' basic-bridge Scheduling destruction of SIP dialog '4368be4a7fc51391404e77d131712de4@195.13.217.88:5060' in 6400 ms (Method: INVITE) == Spawn extension (outgoing-in-city-dial, 00889137529220, 5) exited non-zero on 'SIP/2815-00000005' -- Executing [h@outgoing-in-city-dial:1] GotoIf("SIP/2815-00000005", "1?2:3") in new stack -- Goto (outgoing-in-city-dial,h,2) -- Executing [h@outgoing-in-city-dial:2] Gosub("SIP/2815-00000005", "channel-counter,h,1(sib-83832331290,out,hangup)") in new stack set_destination: Parsing for address/port to send to set_destination: set destination to 77.94.2.34:5060 Reliably Transmitting (no NAT) to 77.94.2.34:5060: BYE sip:89137529220@77.94.2.34:5060 SIP/2.0 Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK4ea5cf13 Max-Forwards: 70 From: "DOMOVOY89137529220" ;tag=as088c8226 To: ;tag=82222164-3859888897-4082892980-1683941628 Call-ID: 4368be4a7fc51391404e77d131712de4@195.13.217.88:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 13.8.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- -- Executing [h@channel-counter:1] Set("SIP/2815-00000005", "TRUNK=sib-83832331290") in new stack -- Executing [h@channel-counter:2] Set("SIP/2815-00000005", "DIR=out") in new stack -- Executing [h@channel-counter:3] Set("SIP/2815-00000005", "ACTION=hangup") in new stack -- Executing [h@channel-counter:4] GotoIf("SIP/2815-00000005", "0?5:7") in new stack -- Goto (channel-counter,h,7) -- Executing [h@channel-counter:7] Set("SIP/2815-00000005", "GROUP(out)=") in new stack -- Executing [h@channel-counter:8] NoOp("SIP/2815-00000005", "Finish if_channel-counter_7") in new stack -- Executing [h@channel-counter:9] Set("SIP/2815-00000005", "ODBC_TRUNK_CHANNELS(sib-83832331290,cur_out)=0") in new stack <--- SIP read from UDP:77.94.2.34:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.13.217.88:5060;branch=z9hG4bK4ea5cf13 From: "DOMOVOY89137529220" ;tag=as088c8226 To: ;tag=82222164-3859888897-4082892980-1683941628 Call-ID: 4368be4a7fc51391404e77d131712de4@195.13.217.88:5060 CSeq: 103 BYE Contact: Server: MERA MVTS3G v.4.4.0-24 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4368be4a7fc51391404e77d131712de4@195.13.217.88:5060' Method: INVITE Reliably Transmitting (no NAT) to 10.10.0.91:5060: OPTIONS sip:2836@10.10.0.91:5060 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK39f94be2 Max-Forwards: 70 From: "asterisk" ;tag=as0ba7216b To: Contact: Call-ID: 3de5aca02b19e8f23db624012f9c6684@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 10.0.11.86:5060: OPTIONS sip:1886@10.0.11.86:5060 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK69688e20 Max-Forwards: 70 From: "asterisk" ;tag=as7ac5fdbe To: Contact: Call-ID: 304493d31fb1e8280839111c37df3197@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.0.11.86:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.111.0.7:5060;received=10.111.0.7;branch=z9hG4bK69688e20 Call-ID: 304493d31fb1e8280839111c37df3197@10.111.0.7:5060 From: "asterisk" ;tag=as7ac5fdbe To: CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, norefersub Allow-Events: presence, refer User-Agent: Domovoy SIP Phone v0.9.0-release/win32 Content-Type: application/sdp Content-Length: 442 v=0 o=- 3669526165 3669526165 IN IP4 10.0.11.86 s=pjmedia c=IN IP4 10.0.11.86 t=0 0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 10.0.11.86 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 19 lines) --- Really destroying SIP dialog '304493d31fb1e8280839111c37df3197@10.111.0.7:5060' Method: OPTIONS <--- SIP read from UDP:10.10.0.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.111.0.7:5060;received=10.111.0.7;branch=z9hG4bK39f94be2 Call-ID: 3de5aca02b19e8f23db624012f9c6684@10.111.0.7:5060 From: "asterisk" ;tag=as0ba7216b To: ;tag=z9hG4bK39f94be2 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/kpml-response+xml, application/kpml-request+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: kpml, dialog, message-summary, refer User-Agent: Cisco-CP3905/9.2.1 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '3de5aca02b19e8f23db624012f9c6684@10.111.0.7:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 10.0.8.82:5062: OPTIONS sip:2831@10.0.8.82:5062 SIP/2.0 Via: SIP/2.0/UDP 10.111.0.7:5060;branch=z9hG4bK32aa9269 Max-Forwards: 70 From: "asterisk" ;tag=as441b50d1 To: Contact: Call-ID: 1fd4f0295f119ed439b9341d4ff71623@10.111.0.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.8.0 Date: Wed, 13 Apr 2016 05:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- dclvccast*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups