Connected to Asterisk 13.35.0 currently running on vultr (pid = 24214) vultr*CLI> vultr*CLI> vultr*CLI> <--- Received SIP request (1116 bytes) from UDP:108.41.213.129:63450 ---> INVITE sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bK4ae79506819F8D25 From: "201" ;tag=CE4796FC-85B373 To: CSeq: 1 INVITE Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Max-Forwards: 70 Content-Type: application/sdp Content-Length: 517 v=0 o=- 1597268412 1597268412 IN IP4 10.10.173.52 s=Polycom IP Phone c=IN IP4 10.10.173.52 t=0 0 a=sendrecv m=audio 2260 RTP/AVP 107 0 9 102 8 127 126 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxptime=20; ptime=20; maxaveragebitrate=24000; cbr=0; useinbandfec=0; usedtx=0 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=rtpmap:126 telephone-event/48000 <--- Transmitting SIP response (512 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bK4ae79506819F8D25 Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc From: "201" ;tag=CE4796FC-85B373 To: ;tag=z9hG4bK4ae79506819F8D25 CSeq: 1 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1597268412/3f480eac6acc7b0267f33c1dc9b9cd79",opaque="2613249a2f6b2373",algorithm=md5,qop="auth" Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Received SIP request (524 bytes) from UDP:108.41.213.129:63450 ---> ACK sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bK4ae79506819F8D25 From: "201" ;tag=CE4796FC-85B373 To: ;tag=z9hG4bK4ae79506819F8D25 CSeq: 1 ACK Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <--- Received SIP request (1407 bytes) from UDP:108.41.213.129:63450 ---> INVITE sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKab21e7007E0E9E87 From: "201" ;tag=CE4796FC-85B373 To: CSeq: 2 INVITE Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Authorization: Digest username="201", realm="asterisk", nonce="1597268412/3f480eac6acc7b0267f33c1dc9b9cd79", qop=auth, cnonce="maXmrJEKN+hZoOx", nc=00000001, opaque="2613249a2f6b2373", uri="sip:205@155.138.217.63:5060;user=phone", response="4d176ee34a01368ec898c1f1fe626dfa", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 517 v=0 o=- 1597268412 1597268412 IN IP4 10.10.173.52 s=Polycom IP Phone c=IN IP4 10.10.173.52 t=0 0 a=sendrecv m=audio 2260 RTP/AVP 107 0 9 102 8 127 126 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxptime=20; ptime=20; maxaveragebitrate=24000; cbr=0; useinbandfec=0; usedtx=0 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=rtpmap:126 telephone-event/48000 == Setting global variable 'SIPDOMAIN' to '155.138.217.63' <--- Transmitting SIP response (332 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKab21e7007E0E9E87 Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc From: "201" ;tag=CE4796FC-85B373 To: CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio CoS mark 5 -- Executing [205@from-test-custom:1] Set("PJSIP/201-00000019", "PJSIP_MEDIA_OFFER(audio)=!all,g722") in new stack -- Executing [205@from-test-custom:2] Set("PJSIP/201-00000019", "PJSIP_SEND_SESSION_REFRESH()=invite") in new stack -- Executing [205@from-test-custom:3] Dial("PJSIP/201-00000019", "PJSIP/205,,b(gosub-jc-custom^s^1)") in new stack -- PJSIP/205-0000001a Internal Gosub(gosub-jc-custom,s,1) start -- Executing [s@gosub-jc-custom:1] Set("PJSIP/205-0000001a", "PJSIP_MEDIA_OFFER(audio)=!all,g722") in new stack -- Executing [s@gosub-jc-custom:2] Set("PJSIP/205-0000001a", "PJSIP_SEND_SESSION_REFRESH()=invite") in new stack -- Executing [s@gosub-jc-custom:3] Return("PJSIP/205-0000001a", "") in new stack == Spawn extension (from-test-custom, 205, 1) exited non-zero on 'PJSIP/205-0000001a' -- PJSIP/205-0000001a Internal Gosub(gosub-jc-custom,s,1) complete GOSUB_RETVAL= -- Called PJSIP/205 == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio CoS mark 5 <--- Transmitting SIP request (963 bytes) to UDP:108.41.213.129:53142 ---> INVITE sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj1c07db0d-e2dd-4fdb-84c3-eff169871cfd From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e To: Contact: Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16111 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "201" Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Type: application/sdp Content-Length: 241 v=0 o=- 1489875390 1489875390 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 15986 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (349 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 100 Trying To: From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16111 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPj1c07db0d-e2dd-4fdb-84c3-eff169871cfd;rport=5060 Server: Cisco/SPA504G-7.6.2f Content-Length: 0 <--- Received SIP response (413 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 180 Ringing To: ;tag=1da4b62d1562271ci1 From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16111 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPj1c07db0d-e2dd-4fdb-84c3-eff169871cfd;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 0 -- PJSIP/205-0000001a is ringing -- PJSIP/205-0000001a is ringing <--- Transmitting SIP response (585 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKab21e7007E0E9E87 Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc From: "201" ;tag=CE4796FC-85B373 To: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE P-Asserted-Identity: "205" Content-Length: 0 <--- Received SIP response (739 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 200 OK To: ;tag=1da4b62d1562271ci1 From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16111 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPj1c07db0d-e2dd-4fdb-84c3-eff169871cfd;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 31974 31974 IN IP4 10.10.173.231 s=- c=IN IP4 10.10.173.231 t=0 0 m=audio 16410 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP request (413 bytes) to UDP:108.41.213.129:53142 ---> ACK sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj438f26bd-02ac-4f14-8417-00afecd6497c From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e To: ;tag=1da4b62d1562271ci1 Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16111 ACK Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Transmitting SIP request (986 bytes) to UDP:108.41.213.129:53142 ---> INVITE sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj6d8abd29-bc4c-49bb-843d-13d4a8bbfeb4 From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e To: ;tag=1da4b62d1562271ci1 Contact: Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16112 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "201" Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Type: application/sdp Content-Length: 241 v=0 o=- 1489875390 1489875391 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 15986 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- PJSIP/205-0000001a answered PJSIP/201-00000019 <--- Transmitting SIP response (951 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKab21e7007E0E9E87 Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc From: "201" ;tag=CE4796FC-85B373 To: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Contact: Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "205" Content-Type: application/sdp Content-Length: 289 v=0 o=- 1597268412 1597268414 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 12262 RTP/AVP 0 9 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- Channel PJSIP/205-0000001a joined 'simple_bridge' basic-bridge <87727a55-85cb-472f-ae56-d8d5848c7db5> -- Channel PJSIP/201-00000019 joined 'simple_bridge' basic-bridge <87727a55-85cb-472f-ae56-d8d5848c7db5> <--- Received SIP request (522 bytes) from UDP:108.41.213.129:63450 ---> ACK sip:155.138.217.63:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKa35e7cc46325B65B From: "201" ;tag=CE4796FC-85B373 To: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd CSeq: 2 ACK Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <--- Transmitting SIP request (1040 bytes) to UDP:108.41.213.129:63450 ---> INVITE sip:201@108.41.213.129:63450 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj007756c5-ada6-43e2-955e-f9ca9ae648a6 From: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd To: "201" ;tag=CE4796FC-85B373 Contact: Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc CSeq: 18121 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "205" Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Type: application/sdp Content-Length: 289 v=0 o=- 1597268412 1597268415 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 12262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (647 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 200 OK To: ;tag=1da4b62d1562271ci1 From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16112 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPj6d8abd29-bc4c-49bb-843d-13d4a8bbfeb4;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 206 Content-Type: application/sdp v=0 o=- 31974 31975 IN IP4 10.10.173.231 s=- c=IN IP4 10.10.173.231 t=0 0 m=audio 16410 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP request (413 bytes) to UDP:108.41.213.129:53142 ---> ACK sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPjee339cdf-851d-469e-a249-ac784d768d02 From: "201" ;tag=1898a1ba-bfae-4553-a603-eba47df14c7e To: ;tag=1da4b62d1562271ci1 Call-ID: 826125b5-4559-4c1a-898f-0093e5f78e6d CSeq: 16112 ACK Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Received SIP request (628 bytes) from UDP:108.41.213.129:53142 ---> NOTIFY sip:155.138.217.63 SIP/2.0 Via: SIP/2.0/UDP 10.10.173.231:5061;branch=z9hG4bK-f98c160d;rport From: ;tag=3afd541e8a3cac1o1 To: Call-ID: d157c3e6-1b02dd99@10.10.173.231 CSeq: 25 NOTIFY Max-Forwards: 70 Authorization: Digest username="205",realm="asterisk",nonce="1597268369/3f64f904a108856162c9e9f19c72a450",uri="sip:155.138.217.63",algorithm=MD5,response="5089b58058f3d6bb6d4183c567f823e4",opaque="724509af4d97f860",qop=auth,nc=00000004,cnonce="4917d4b2" Contact: Event: keep-alive User-Agent: Cisco/SPA504G-7.6.2f Content-Length: 0 <--- Transmitting SIP response (496 bytes) to UDP:108.41.213.129:53142 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.173.231:5061;rport=53142;received=108.41.213.129;branch=z9hG4bK-f98c160d Call-ID: d157c3e6-1b02dd99@10.10.173.231 From: ;tag=3afd541e8a3cac1o1 To: ;tag=z9hG4bK-f98c160d CSeq: 25 NOTIFY WWW-Authenticate: Digest realm="asterisk",nonce="1597268414/a790142adaef48323355e30467eabd61",opaque="447f9af32b50e146",stale=true,algorithm=md5,qop="auth" Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Received SIP request (628 bytes) from UDP:108.41.213.129:53142 ---> NOTIFY sip:155.138.217.63 SIP/2.0 Via: SIP/2.0/UDP 10.10.173.231:5061;branch=z9hG4bK-df5fbf86;rport From: ;tag=3afd541e8a3cac1o1 To: Call-ID: d157c3e6-1b02dd99@10.10.173.231 CSeq: 26 NOTIFY Max-Forwards: 70 Authorization: Digest username="205",realm="asterisk",nonce="1597268414/a790142adaef48323355e30467eabd61",uri="sip:155.138.217.63",algorithm=MD5,response="f2c7df28f93319415d88bfa1f190de04",opaque="447f9af32b50e146",qop=auth,nc=00000001,cnonce="f62f4943" Contact: Event: keep-alive User-Agent: Cisco/SPA504G-7.6.2f Content-Length: 0 <--- Transmitting SIP response (342 bytes) to UDP:108.41.213.129:53142 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.10.173.231:5061;rport=53142;received=108.41.213.129;branch=z9hG4bK-df5fbf86 Call-ID: d157c3e6-1b02dd99@10.10.173.231 From: ;tag=3afd541e8a3cac1o1 To: ;tag=z9hG4bK-df5fbf86 CSeq: 26 NOTIFY Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Received SIP response (833 bytes) from UDP:108.41.213.129:63450 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj007756c5-ada6-43e2-955e-f9ca9ae648a6 From: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd To: "201" ;tag=CE4796FC-85B373 CSeq: 18121 INVITE Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Supported: replaces,100rel User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Allow-Events: conference,talk,hold Accept-Language: en Content-Type: application/sdp Content-Length: 211 v=0 o=- 1597268412 1597268413 IN IP4 10.10.173.52 s=Polycom IP Phone c=IN IP4 10.10.173.52 t=0 0 a=sendrecv m=audio 2260 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=sendrecv <--- Transmitting SIP request (417 bytes) to UDP:108.41.213.129:63450 ---> ACK sip:201@108.41.213.129:63450 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPjaa890386-0ebc-4899-8592-dcc9d83834af From: ;tag=edc49565-d6b2-4250-8938-e6c611d2bcbd To: "201" ;tag=CE4796FC-85B373 Call-ID: a97ebacb104d55cd18d28fd1f5bdb6bc CSeq: 18121 ACK Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Length: 0 vultr*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups [root@vultr freepbx]#