localhost*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sbc5.fr.sip.ovh:5060 N 003397266511 1785 Registered Tue, 11 Jan 2022 16:13:35 1 SIP registrations. localhost*CLI> sip set debug on SIP Debugging enabled Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK52a34291 Max-Forwards: 70 From: "asterisk" ;tag=as0430fd8f To: Contact: Call-ID: 59103ab749f40adc3886a4f82cf9cf7e@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:20:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK52a34291 Contact: To: ;tag=b110ce2e From: "asterisk" ;tag=as0430fd8f Call-ID: 59103ab749f40adc3886a4f82cf9cf7e@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '59103ab749f40adc3886a4f82cf9cf7e@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Reliably Transmitting (no NAT) to 91.121.129.132:5060: OPTIONS sip:sbc5.fr.sip.ovh SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK40421864 Max-Forwards: 70 From: "asterisk" ;tag=as0e0785e6 To: Contact: Call-ID: 6f42b93430b5948d034814fe38dd10c8@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:20:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-06947-100619271-64fa60f0@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-06947-100619270-3e0fff30 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-UFJZ-a40f17be-1ff2012d Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-UFJZ-a40f17be-1ff2012d;received=91.121.129.132;rport=5060 From: ;tag=00-06947-100619270-3e0fff30 To: ;tag=as1d808bd5 Call-ID: 00-06947-100619271-64fa60f0@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-06947-100619271-64fa60f0@91.121.129.132' in 32000 ms (Method: OPTIONS) <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 6f42b93430b5948d034814fe38dd10c8@82.165.120.235:16060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0e0785e6 To: ;tag=00-32031-1006192f9-59e54c12 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK40421864 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '6f42b93430b5948d034814fe38dd10c8@82.165.120.235:16060' Method: OPTIONS Really destroying SIP dialog '00-03310-100614032-4b289d86@91.121.129.132' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---5965100d635cdf0c;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1343 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="393063b8",uri="sip:82.165.120.235",response="5e7d871da9b2f69df0039febb6be0a38",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Sending to 82.66.71.140:60295 (no NAT) <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---5965100d635cdf0c;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as5f25a364 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1343 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7189a74c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---2d344e32f3d64048;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1344 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="7189a74c",uri="sip:82.165.120.235",response="d83ca904770673165eb0693469f745b7",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK1b88f5f0 Max-Forwards: 70 From: "asterisk" ;tag=as24a200c1 To: Contact: Call-ID: 18d52f961729eafd18db089e5e5f2d11@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:21:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---2d344e32f3d64048;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as5f25a364 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1344 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 90 Contact: ;expires=90 Date: Tue, 11 Jan 2022 16:21:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c0d6e426ed6a3a577080c46381c0d90@82.165.120.235:16060' in 7232 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 82.66.71.140:60295: NOTIFY sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK483ba306 Max-Forwards: 70 From: "asterisk" ;tag=as295f5b08 To: Contact: Call-ID: 3c0d6e426ed6a3a577080c46381c0d90@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 100 Messages-Waiting: no Message-Account: sip:asterisk@82.165.120.235:16060 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK1b88f5f0 Contact: To: ;tag=4d33725e From: "asterisk" ;tag=as24a200c1 Call-ID: 18d52f961729eafd18db089e5e5f2d11@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK483ba306 Contact: To: ;tag=af20c86f From: "asterisk" ;tag=as295f5b08 Call-ID: 3c0d6e426ed6a3a577080c46381c0d90@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '18d52f961729eafd18db089e5e5f2d11@82.165.120.235:16060' Method: OPTIONS Really destroying SIP dialog '3c0d6e426ed6a3a577080c46381c0d90@82.165.120.235:16060' Method: NOTIFY <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-00387-10061e416-522dd4d2@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-00387-10061e415-547bc099 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-JVNO-a40f4dae-29e79987 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-JVNO-a40f4dae-29e79987;received=91.121.129.132;rport=5060 From: ;tag=00-00387-10061e415-547bc099 To: ;tag=as6ef169d1 Call-ID: 00-00387-10061e416-522dd4d2@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-00387-10061e416-522dd4d2@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-06947-100619271-64fa60f0@91.121.129.132' Method: OPTIONS <--- SIP read from UDP:91.121.129.132:5060 ---> INVITE sip:s@82.165.120.235:16060;transport=udp SIP/2.0 Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: Content-Type: application/sdp CSeq: 384038943 INVITE From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Max-Forwards: 29 Record-Route: ;session=231285 To: Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-VNLN-a40f6040-627d04c9 Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 320 v=0 o=anonymous 164191809248 164191809248 IN IP4 91.121.129.132 s=SIP Call c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 8 0 18 96 b=AS:82 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 15 lines) --- Sending to 91.121.129.132:5060 (no NAT) Sending to 91.121.129.132:5060 (no NAT) Using INVITE request as basis request - 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Found peer 'atmos-default-context' for '0671760272' from 91.121.129.132:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 96 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x564d9d54bf30 -- Strict RTP learning after remote address set to: 91.121.129.154:30642 Peer audio RTP is at port 91.121.129.154:30642 Looking for s in atmos (domain 82.165.120.235) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-VNLN-a40f6040-627d04c9;received=91.121.129.132 Record-Route: ;session=231285 From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 To: Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 384038943 INVITE Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [s@atmos:1] Set("SIP/atmos-default-context-00000052", "ATMOS_CONTEXT=START") in new stack -- Executing [s@atmos:2] Set("SIP/atmos-default-context-00000052", "ATMOS_INPUT=-") in new stack -- Executing [s@atmos:3] Goto("SIP/atmos-default-context-00000052", "atmos-server-query,s,1") in new stack -- Goto (atmos-server-query,s,1) -- Executing [s@atmos-server-query:1] AGI("SIP/atmos-default-context-00000052", "atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/STA) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/atmos-request.agi atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: MEUHHHHHH atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: https://atmos-app.clangen.com:5218/asterisk-call/91f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/- atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: {userProps:{},context:atmos-play-audio,exten:s,pridd6-9de5-fe94a37b018b,ttsMessage:,delay:0,dialSip:} atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: JSON : atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: {'userProps': {}, 'context': 'atmos-play-audio', 'dioFile': 'de785e74-b88b-4dd6-9de5-fe94a37b018b', 'ttsMessage': '', 'delay': 0, 'dialSip': ''} -- AGI Script atmos-request.agi completed, returning 0 -- Executing [s@atmos-server-query:2] Verbose("SIP/atmos-default-context-00000052", ""Nous avons un Goto(atmos-play-audio, s, 1) avec Audio 'de785e74-b88b-4dd6-9de5-fe94a37b018b' et TTS ''"") in new stack Nous avons un Goto(atmos-play-audio, s, 1) avec Audio 'de785e74-b88b-4dd6-9de5-fe94a37b018b' et TTS '' -- Executing [s@atmos-server-query:3] Goto("SIP/atmos-default-context-00000052", "atmos-play-audio,s,1") in new stack -- Goto (atmos-play-audio,s,1) -- Executing [s@atmos-play-audio:1] BackGround("SIP/atmos-default-context-00000052", "/home/audio/de785e74-b88b-4dd6-9de5-fe94a37b018b") in new stack Audio is at 12888 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-VNLN-a40f6040-627d04c9;received=91.121.129.132 Record-Route: ;session=231285 From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 To: ;tag=as569c000c Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 384038943 INVITE Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 401889503 401889503 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.165.120.235 t=0 0 m=audio 12888 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:91.121.129.132:5060 ---> ACK sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: CSeq: 384038943 ACK From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Max-Forwards: 29 To: ;tag=as569c000c Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-GSXI-a40f60f6-61714b5e User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:91.121.129.132:5060 ---> INVITE sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: Content-Type: application/sdp CSeq: 384038944 INVITE From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Max-Forwards: 29 Record-Route: ;session=231285 To: ;tag=as569c000c Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-SJSP-a40f6118-574b7334 Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 245 v=0 o=anonymous 164191809248 164191809249 IN IP4 91.121.129.132 s=SIP Call c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 0 96 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 12 lines) --- Sending to 91.121.129.132:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 96 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 96 Capabilities: us - (ulaw|alaw), 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|) > 0x564d9d54bf30 -- Strict RTP learning after remote address set to: 91.121.129.154:30642 Peer audio RTP is at port 91.121.129.154:30642 <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-SJSP-a40f6118-574b7334;received=91.121.129.132 Record-Route: ;session=231285 From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 To: ;tag=as569c000c Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 384038944 INVITE Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 12888 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-SJSP-a40f6118-574b7334;received=91.121.129.132 Record-Route: ;session=231285 From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 To: ;tag=as569c000c Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 384038944 INVITE Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 401889503 401889504 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.165.120.235 t=0 0 m=audio 12888 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:91.121.129.132:5060 ---> ACK sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: CSeq: 384038944 ACK From: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Max-Forwards: 29 To: ;tag=as569c000c Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-HLTV-a40f611b-4c868398 User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Playing '/home/audio/de785e74-b88b-4dd6-9de5-fe94a37b018b.slin' (language 'fr') > 0x564d9d54bf30 -- Strict RTP switching to RTP target address 91.121.129.154:30642 as source <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> > 0x564d9d54bf30 -- Strict RTP learning complete - Locking on source address 91.121.129.154:30642 -- Executing [s@atmos-play-audio:2] Goto("SIP/atmos-default-context-00000052", "atmos-server-query,s,1") in new stack -- Goto (atmos-server-query,s,1) -- Executing [s@atmos-server-query:1] AGI("SIP/atmos-default-context-00000052", "atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/STA) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/atmos-request.agi Really destroying SIP dialog 'j1QnMdRDShopudQi0xToPA..' Method: REGISTER atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: MEUHHHHHH atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: https://atmos-app.clangen.com:5218/asterisk-call/91f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/- atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: {userProps:{},context:atmos-pick-up,exten:s,priori:60,dialSip:SIP/Guillaume} atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: JSON : atmos-request.agi, https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/info/START/0671760272/s/1641918092.136/-: {'userProps': {}, 'context': 'atmos-pick-up', 'extFile': '', 'ttsMessage': '', 'delay': 60, 'dialSip': 'SIP/Guillaume'} -- AGI Script atmos-request.agi completed, returning 0 -- Executing [s@atmos-server-query:2] Verbose("SIP/atmos-default-context-00000052", ""Nous avons un Goto(atmos-pick-up, s, 1) avec Audio '' et TTS ''"") in new stack Nous avons un Goto(atmos-pick-up, s, 1) avec Audio '' et TTS '' -- Executing [s@atmos-server-query:3] Goto("SIP/atmos-default-context-00000052", "atmos-pick-up,s,1") in new stack -- Goto (atmos-pick-up,s,1) -- Executing [s@atmos-pick-up:1] Verbose("SIP/atmos-default-context-00000052", ""PickUp avec SIP/Guillaume pour une durée de 60"") in new stack PickUp avec SIP/Guillaume pour une durée de 60 -- Executing [s@atmos-pick-up:2] Dial("SIP/atmos-default-context-00000052", "SIP/Guillaume,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 17982 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 82.66.71.140:60295: INVITE sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2676bb57 Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:21:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 290 v=0 o=root 1759593079 1759593079 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.165.120.235 t=0 0 m=audio 17982 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/Guillaume Retransmitting #1 (no NAT) to 82.66.71.140:60295: INVITE sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2676bb57 Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:21:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 290 v=0 o=root 1759593079 1759593079 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.165.120.235 t=0 0 m=audio 17982 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2676bb57 Contact: ;+sip.instance="" To: ;tag=47ffc365 From: "+33671760272" ;tag=as301eaff6 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 INVITE User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/Guillaume-00000053 is ringing <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2676bb57 Contact: ;+sip.instance="" To: ;tag=47ffc365 From: "+33671760272" ;tag=as301eaff6 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 INVITE User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/Guillaume-00000053 is ringing <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2676bb57 Contact: ;+sip.instance="" To: ;tag=47ffc365 From: "+33671760272" ;tag=as301eaff6 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Content-Type: application/sdp Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 231 v=0 o=- 1641918101 1 IN IP4 82.66.71.140 s=ps c=IN IP4 82.66.71.140 t=0 0 m=audio 10012 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=mid:audio a=sendrecv <-------------> --- (13 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7f67dc0292e0 -- Strict RTP learning after remote address set to: 82.66.71.140:10012 Peer audio RTP is at port 82.66.71.140:10012 sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 82.66.71.140:60295 Transmitting (no NAT) to 82.66.71.140:60295: ACK sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK30f512f8 Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: ;tag=47ffc365 Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Content-Length: 0 --- -- SIP/Guillaume-00000053 answered SIP/atmos-default-context-00000052 -- Channel SIP/Guillaume-00000053 joined 'simple_bridge' basic-bridge <1992bab6-81f7-4c3d-81f1-39419e315ff3> -- Channel SIP/atmos-default-context-00000052 joined 'simple_bridge' basic-bridge <1992bab6-81f7-4c3d-81f1-39419e315ff3> > Bridge 1992bab6-81f7-4c3d-81f1-39419e315ff3: switching from simple_bridge technology to native_rtp set_destination: Parsing for address/port to send to set_destination: set destination to 91.121.129.132:5060 Audio is at 12888 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 91.121.129.132:5060: INVITE sip:91.121.129.132:5060 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK49441a26 Route: Max-Forwards: 70 From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Contact: Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 262 v=0 o=root 401889503 401889505 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.66.71.140 t=0 0 m=audio 10012 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- set_destination: Parsing for address/port to send to set_destination: set destination to 82.66.71.140:60295 Audio is at 17982 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 82.66.71.140:60295: INVITE sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK637cb13a Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: ;tag=47ffc365 Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 1759593079 1759593080 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- > Remotely bridged 'SIP/atmos-default-context-00000052' and 'SIP/Guillaume-00000053' - media will flow directly between them <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 100 Trying Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 102 INVITE From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK49441a26 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: Content-Type: application/sdp CSeq: 102 INVITE From: ;tag=as569c000c Record-Route: ;session=231285 To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK49441a26 Allow: UPDATE,REFER,INFO Server: Cirpack/v4.76 (gw_sip) Content-Length: 245 v=0 o=anonymous 164191809248 164191809249 IN IP4 91.121.129.132 s=SIP Call c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 0 96 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 91.121.129.132:5060 Transmitting (no NAT) to 91.121.129.132:5060: ACK sip:91.121.129.132:5060 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK70583129 Route: Max-Forwards: 70 From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Contact: Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 102 ACK User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Content-Length: 0 --- > 0x7f67dc0292e0 -- Strict RTP qualifying stream type: audio Retransmitting #1 (no NAT) to 82.66.71.140:60295: INVITE sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK637cb13a Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: ;tag=47ffc365 Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 103 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 1759593079 1759593080 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK637cb13a Contact: ;+sip.instance="" To: ;tag=47ffc365 From: "+33671760272" ;tag=as301eaff6 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Content-Type: application/sdp Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 231 v=0 o=- 1641918101 2 IN IP4 82.66.71.140 s=ps c=IN IP4 82.66.71.140 t=0 0 m=audio 10012 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=mid:audio a=sendrecv <-------------> --- (13 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7f67dc0292e0 -- Strict RTP learning after remote address set to: 82.66.71.140:10012 Peer audio RTP is at port 82.66.71.140:10012 set_destination: Parsing for address/port to send to set_destination: set destination to 82.66.71.140:60295 Transmitting (no NAT) to 82.66.71.140:60295: ACK sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK68bdef20 Max-Forwards: 70 From: "+33671760272" ;tag=as301eaff6 To: ;tag=47ffc365 Contact: Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 103 ACK User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Content-Length: 0 --- <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Reliably Transmitting (no NAT) to 91.121.129.132:5060: OPTIONS sip:sbc5.fr.sip.ovh SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK59524991 Max-Forwards: 70 From: "asterisk" ;tag=as3a7ae7a5 To: Contact: Call-ID: 6e64f6784fd8c40d4be692c26df7a0e0@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:21:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 6e64f6784fd8c40d4be692c26df7a0e0@82.165.120.235:16060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as3a7ae7a5 To: ;tag=00-31184-10062350d-69491668 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK59524991 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '6e64f6784fd8c40d4be692c26df7a0e0@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-06399-100623604-773a0268@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-06399-100623603-1ff6a6a1 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-GYZR-a40f86b7-2cd4cf90 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-GYZR-a40f86b7-2cd4cf90;received=91.121.129.132;rport=5060 From: ;tag=00-06399-100623603-1ff6a6a1 To: ;tag=as7bb74060 Call-ID: 00-06399-100623604-773a0268@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-06399-100623604-773a0268@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-00387-10061e416-522dd4d2@91.121.129.132' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK69e807d7 Max-Forwards: 70 From: "asterisk" ;tag=as50197295 To: Contact: Call-ID: 634c85d70f1572fc14d473014358da1e@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:22:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK69e807d7 Contact: To: ;tag=e963ca2b From: "asterisk" ;tag=as50197295 Call-ID: 634c85d70f1572fc14d473014358da1e@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '634c85d70f1572fc14d473014358da1e@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-02848-1006287ea-5ba73015@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-02848-1006287e9-69bb051a Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-ULLI-a40fbc91-067d5047 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-ULLI-a40fbc91-067d5047;received=91.121.129.132;rport=5060 From: ;tag=00-02848-1006287e9-69bb051a To: ;tag=as7ea0bd2d Call-ID: 00-02848-1006287ea-5ba73015@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-02848-1006287ea-5ba73015@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-06399-100623604-773a0268@91.121.129.132' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---b295de7aca4f3d7a;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1345 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="7189a74c",uri="sip:82.165.120.235",response="d83ca904770673165eb0693469f745b7",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Sending to 82.66.71.140:60295 (no NAT) <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---b295de7aca4f3d7a;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as13ad1f4c Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1345 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ffee50d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---829b48650ff0c212;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1346 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="2ffee50d",uri="sip:82.165.120.235",response="1ec26d51053e35212b31cef9e5e3c974",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK487a5175 Max-Forwards: 70 From: "asterisk" ;tag=as110b6cfc To: Contact: Call-ID: 6d05637952e25030575d82a86f5eac86@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:22:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---829b48650ff0c212;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as13ad1f4c Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1346 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 90 Contact: ;expires=90 Date: Tue, 11 Jan 2022 16:22:30 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '55a7910350c5ffd55108189d0e35e291@82.165.120.235:16060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 82.66.71.140:60295: NOTIFY sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK4f888cfc Max-Forwards: 70 From: "asterisk" ;tag=as6b2ec603 To: Contact: Call-ID: 55a7910350c5ffd55108189d0e35e291@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 100 Messages-Waiting: no Message-Account: sip:asterisk@82.165.120.235:16060 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK487a5175 Contact: To: ;tag=5c65690b From: "asterisk" ;tag=as110b6cfc Call-ID: 6d05637952e25030575d82a86f5eac86@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK4f888cfc Contact: To: ;tag=b4968c39 From: "asterisk" ;tag=as6b2ec603 Call-ID: 55a7910350c5ffd55108189d0e35e291@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '6d05637952e25030575d82a86f5eac86@82.165.120.235:16060' Method: OPTIONS Really destroying SIP dialog '55a7910350c5ffd55108189d0e35e291@82.165.120.235:16060' Method: NOTIFY <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Reliably Transmitting (no NAT) to 91.121.129.132:5060: OPTIONS sip:sbc5.fr.sip.ovh SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK3f8f98e7 Max-Forwards: 70 From: "asterisk" ;tag=as6e7be557 To: Contact: Call-ID: 4c9d1e174410514b75c1f5600f43c1c9@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:22:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 4c9d1e174410514b75c1f5600f43c1c9@82.165.120.235:16060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as6e7be557 To: ;tag=00-30329-10062d8f5-34423cbb Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK3f8f98e7 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '4c9d1e174410514b75c1f5600f43c1c9@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-00851-10062d9e1-2f184a2f@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-00851-10062d9e0-13103b46 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-KZQP-a40ff803-0f428464 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-KZQP-a40ff803-0f428464;received=91.121.129.132;rport=5060 From: ;tag=00-00851-10062d9e0-13103b46 To: ;tag=as3240aa36 Call-ID: 00-00851-10062d9e1-2f184a2f@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-00851-10062d9e1-2f184a2f@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-02848-1006287ea-5ba73015@91.121.129.132' Method: OPTIONS Really destroying SIP dialog 'j1QnMdRDShopudQi0xToPA..' Method: REGISTER <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:82.66.71.140:60295 ---> BYE sip:0671760272@82.165.120.235:16060 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---7db73e211af30370;rport Max-Forwards: 70 Contact: ;+sip.instance="" To: "+33671760272";tag=as301eaff6 From: ;tag=47ffc365 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 2 BYE User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Scheduling destruction of SIP dialog '1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---7db73e211af30370;received=82.66.71.140;rport=60295 From: ;tag=47ffc365 To: "+33671760272";tag=as301eaff6 Call-ID: 1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060 CSeq: 2 BYE Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/Guillaume-00000053 left 'native_rtp' basic-bridge <1992bab6-81f7-4c3d-81f1-39419e315ff3> set_destination: Parsing for address/port to send to set_destination: set destination to 91.121.129.132:5060 Audio is at 12888 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 91.121.129.132:5060: INVITE sip:91.121.129.132:5060 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK6b8b53c8 Route: Max-Forwards: 70 From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Contact: Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 103 INVITE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 264 v=0 o=root 401889503 401889506 IN IP4 82.165.120.235 s=Asterisk PBX 16.2.1~dfsg-2ubuntu1 c=IN IP4 82.165.120.235 t=0 0 m=audio 12888 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Channel SIP/atmos-default-context-00000052 left 'native_rtp' basic-bridge <1992bab6-81f7-4c3d-81f1-39419e315ff3> == Spawn extension (atmos-pick-up, s, 2) exited non-zero on 'SIP/atmos-default-context-00000052' -- Executing [h@atmos-pick-up:1] Goto("SIP/atmos-default-context-00000052", "atmos-caller-hangup,s,1") in new stack -- Goto (atmos-caller-hangup,s,1) -- Executing [s@atmos-caller-hangup:1] AGI("SIP/atmos-default-context-00000052", "atmos-request.agi,https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/hangup/S-") in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/atmos-request.agi <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 100 Trying Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 103 INVITE From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK6b8b53c8 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: Content-Type: application/sdp CSeq: 103 INVITE From: ;tag=as569c000c Record-Route: ;session=231285 To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK6b8b53c8 Allow: UPDATE,REFER,INFO Server: Cirpack/v4.76 (gw_sip) Content-Length: 245 v=0 o=anonymous 164191809248 164191809249 IN IP4 91.121.129.132 s=SIP Call c=IN IP4 91.121.129.154 t=0 0 m=audio 30642 RTP/AVP 0 96 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 91.121.129.132:5060 Transmitting (no NAT) to 91.121.129.132:5060: ACK sip:91.121.129.132:5060 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK725d53f6 Route: Max-Forwards: 70 From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Contact: Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 103 ACK User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Content-Length: 0 --- atmos-request.agi,https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/hangup/START/0671760272/s/1641918092.136/-: MEUHHHHHH atmos-request.agi,https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/hangup/START/0671760272/s/1641918092.136/-: https://atmos-app.clangen.com:5218/asterisk-call/91f0fc4d0eae4c164221c46d1e/hangup/START/0671760272/s/1641918092.136/- atmos-request.agi,https://atmos-app.clangen.com:5218/asterisk-call/q/5f4158f90327605b58e922aa/060f7191f0fc4d0eae4c164221c46d1e/hangup/START/0671760272/s/1641918092.136/-: -- AGI Script atmos-request.agi completed, returning 0 Scheduling destruction of SIP dialog '28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 91.121.129.132:5060 Reliably Transmitting (no NAT) to 91.121.129.132:5060: BYE sip:91.121.129.132:5060 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK18541029 Route: Max-Forwards: 70 From: ;tag=as569c000c To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh CSeq: 104 BYE User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh Contact: CSeq: 104 BYE From: ;tag=as569c000c Record-Route: ;session=231285 To: "+33671760272" ;tag=28086-AM-1553aa02-54db65205 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK18541029 Server: Cirpack/v4.76 (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '28086-UF-1553aa01-3a5a5eac6@sbc5.fr.sip.ovh' Method: ACK <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Really destroying SIP dialog '1947876c0c6cae9f2e61b6fa19e20ad8@82.165.120.235:16060' Method: BYE <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-06471-100632b92-554d8bc5@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-06471-100632b91-04118e03 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-HBNL-a4102f3f-65fcff8a Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-HBNL-a4102f3f-65fcff8a;received=91.121.129.132;rport=5060 From: ;tag=00-06471-100632b91-04118e03 To: ;tag=as0a8b855d Call-ID: 00-06471-100632b92-554d8bc5@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-06471-100632b92-554d8bc5@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-00851-10062d9e1-2f184a2f@91.121.129.132' Method: OPTIONS Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK4079b1b9 Max-Forwards: 70 From: "asterisk" ;tag=as52cd538c To: Contact: Call-ID: 0909177d2066528a58aa68c709d6297b@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:23:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK4079b1b9 Contact: To: ;tag=a05b2178 From: "asterisk" ;tag=as52cd538c Call-ID: 0909177d2066528a58aa68c709d6297b@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '0909177d2066528a58aa68c709d6297b@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---940c862340318f19;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1347 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="2ffee50d",uri="sip:82.165.120.235",response="1ec26d51053e35212b31cef9e5e3c974",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Sending to 82.66.71.140:60295 (no NAT) <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---940c862340318f19;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as08b50dbb Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1347 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d2c0bdc" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> REGISTER sip:82.165.120.235 SIP/2.0 Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---5194784cc9b9ee73;rport Max-Forwards: 70 Contact: ;+sip.instance="";reg-id=1 To: From: ;tag=fd254a11 Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1348 REGISTER Expires: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Authorization: Digest username="Guillaume",realm="asterisk",nonce="5d2c0bdc",uri="sip:82.165.120.235",response="512e66d423a490f2fd1fb47f7ea015e9",algorithm=MD5 Allow-Events: hold, talk, conference, dialog x-p-push: device-os=android;device-uid=d1dpSsZ7gjI:APA91bFgYVNuGi_6lgmE1T_fAlNyzDWXhHVHjBC3JVtVwvI9k_TaLO6qkB5cpljFpnaEPUVw3JUXYV7NF72nYlQsMOdGdq9SoN1CilHwZxC-xFRzDSr5gqt2ykom7t5eBe4A4tNwT3-y;allow-call-push=true;allow-mep.portgo Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Sending to 82.66.71.140:60295 (no NAT) Reliably Transmitting (no NAT) to 82.66.71.140:60295: OPTIONS sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2b80ba96 Max-Forwards: 70 From: "asterisk" ;tag=as6f1a3314 To: Contact: Call-ID: 016685bf6e96f982618450731fdac406@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:23:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.11.218:13852;branch=z9hG4bK-524287-1---5194784cc9b9ee73;received=82.66.71.140;rport=60295 From: ;tag=fd254a11 To: ;tag=as08b50dbb Call-ID: j1QnMdRDShopudQi0xToPA.. CSeq: 1348 REGISTER Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 90 Contact: ;expires=90 Date: Tue, 11 Jan 2022 16:23:51 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '66e71316620cf2c12689b1c47a55cf68@82.165.120.235:16060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 82.66.71.140:60295: NOTIFY sip:Guillaume@82.66.71.140:60295 SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK6930616b Max-Forwards: 70 From: "asterisk" ;tag=as319adb49 To: Contact: Call-ID: 66e71316620cf2c12689b1c47a55cf68@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 100 Messages-Waiting: no Message-Account: sip:asterisk@82.165.120.235:16060 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog 'j1QnMdRDShopudQi0xToPA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2b80ba96 Contact: To: ;tag=3ba7d630 From: "asterisk" ;tag=as6f1a3314 Call-ID: 016685bf6e96f982618450731fdac406@82.165.120.235:16060 CSeq: 102 OPTIONS Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: PortSIP UC Client Android - v10.9.1 Allow-Events: hold, talk, conference, dialog Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:82.66.71.140:60295 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK6930616b Contact: To: ;tag=8c5ef868 From: "asterisk" ;tag=as319adb49 Call-ID: 66e71316620cf2c12689b1c47a55cf68@82.165.120.235:16060 CSeq: 102 NOTIFY User-Agent: PortSIP UC Client Android - v10.9.1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '016685bf6e96f982618450731fdac406@82.165.120.235:16060' Method: OPTIONS Really destroying SIP dialog '66e71316620cf2c12689b1c47a55cf68@82.165.120.235:16060' Method: NOTIFY <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> Reliably Transmitting (no NAT) to 91.121.129.132:5060: OPTIONS sip:sbc5.fr.sip.ovh SIP/2.0 Via: SIP/2.0/UDP 82.165.120.235:16060;branch=z9hG4bK2f410de5 Max-Forwards: 70 From: "asterisk" ;tag=as5131eed3 To: Contact: Call-ID: 2b6c910a2276c164617e761964c06f3b@82.165.120.235:16060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1 Date: Tue, 11 Jan 2022 16:23:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:91.121.129.132:5060 ---> SIP/2.0 200 OK Call-ID: 2b6c910a2276c164617e761964c06f3b@82.165.120.235:16060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as5131eed3 To: ;tag=00-32752-100637c4d-3aded632 Via: SIP/2.0/UDP 82.165.120.235:16060;received=82.165.120.235;rport=16060;branch=z9hG4bK2f410de5 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '2b6c910a2276c164617e761964c06f3b@82.165.120.235:16060' Method: OPTIONS <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-01311-100637d39-6b430694@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-01311-100637d38-4fd282eb Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-VULH-a4106aa0-024c1cd2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-VULH-a4106aa0-024c1cd2;received=91.121.129.132;rport=5060 From: ;tag=00-01311-100637d38-4fd282eb To: ;tag=as0023de84 Call-ID: 00-01311-100637d39-6b430694@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-01311-100637d39-6b430694@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '00-06471-100632b92-554d8bc5@91.121.129.132' Method: OPTIONS <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:82.66.71.140:60295 ---> <-------------> <--- SIP read from UDP:91.121.129.132:5060 ---> OPTIONS sip:s@82.165.120.235:16060 SIP/2.0 Call-ID: 00-05419-10063cf31-3e613228@91.121.129.132 Contact: CSeq: 1 OPTIONS From: ;tag=00-05419-10063cf30-0c166bf9 Max-Forwards: 70 To: Via: SIP/2.0/UDP 91.121.129.132:5060;rport;branch=z9hG4bK-NUTH-a410a21e-5172c408 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 91.121.129.132:5060 (no NAT) Looking for s in atmos-default-context (domain 82.165.120.235) <--- Transmitting (no NAT) to 91.121.129.132:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.129.132:5060;branch=z9hG4bK-NUTH-a410a21e-5172c408;received=91.121.129.132;rport=5060 From: ;tag=00-05419-10063cf30-0c166bf9 To: ;tag=as03a3a467 Call-ID: 00-05419-10063cf31-3e613228@91.121.129.132 CSeq: 1 OPTIONS Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1 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 '00-05419-10063cf31-3e613228@91.121.129.132' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog 'j1QnMdRDShopudQi0xToPA..' Method: REGISTER Really destroying SIP dialog '00-01311-100637d39-6b430694@91.121.129.132' Method: OPTIONS localhost*CLI> ~~~