SIP/2.0 200 OK From: "asterisk";tag=as1800f1ad To: ;tag=5eb628-b404a8c0-13c4-5d2db0f5-5776136e-5d2db0f5 Call-ID: 4660bae017c1593824f99c813ec3365e@13.234.154.122:5060 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK69d84710 Supported: replaces Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '4660bae017c1593824f99c813ec3365e@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:1367 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 Via: SIP/2.0/UDP 103.249.83.162:1367;branch=z9hG4bKcd13ee31864BDAE0 From: "108" ;tag=17246520-564ACD05 To: CSeq: 662 REGISTER Call-ID: 3a5aebf8-7f6334d-36b45694@192.168.4.132 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Authorization: Digest username="108", realm="asterisk", nonce="548fd1d7", uri="sip:bmw.pulsework360.com", response="f9d0669240dfeeced99f5503cfa8f760", algorithm=MD5 Max-Forwards: 70 Expires: 10 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 103.249.83.162:1367 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:1367: OPTIONS sip:108@103.249.83.162:1367 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK0a86385c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3745c600 To: Contact: Call-ID: 44f6c7d40c1be2395f6ca72c3947ac81@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:1367 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:1367;branch=z9hG4bKcd13ee31864BDAE0;received=103.249.83.162;rport=1367 From: "108" ;tag=17246520-564ACD05 To: ;tag=as440c371e Call-ID: 3a5aebf8-7f6334d-36b45694@192.168.4.132 CSeq: 662 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:12 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5683a1582ba3674778d423fa76d033db@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:1367: NOTIFY sip:108@103.249.83.162:1367 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK0aa4dab5;rport Max-Forwards: 70 From: "asterisk" ;tag=as02efc783 To: Contact: Call-ID: 5683a1582ba3674778d423fa76d033db@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '3a5aebf8-7f6334d-36b45694@192.168.4.132' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:1367 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK0a86385c;rport From: "asterisk" ;tag=as3745c600 To: "108" ;tag=AFA9603C-29B7DAB9 CSeq: 102 OPTIONS Call-ID: 44f6c7d40c1be2395f6ca72c3947ac81@13.234.154.122:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '44f6c7d40c1be2395f6ca72c3947ac81@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:1367 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK0aa4dab5;rport From: "asterisk" ;tag=as02efc783 To: "108" ;tag=47B6768D-58B9A254 CSeq: 102 NOTIFY Call-ID: 5683a1582ba3674778d423fa76d033db@13.234.154.122:5060 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '5683a1582ba3674778d423fa76d033db@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:1369 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 Via: SIP/2.0/UDP 103.249.83.162:1369;branch=z9hG4bKec24a1a7B072E84D From: "109" ;tag=F83AF917-267B271D To: CSeq: 705 REGISTER Call-ID: d901a7a7-bfa24a4d-6dc3158f@192.168.4.128 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Authorization: Digest username="109", realm="asterisk", nonce="105764d4", uri="sip:bmw.pulsework360.com", response="d0b400885b7c87c120569ead181f0ad2", algorithm=MD5 Max-Forwards: 70 Expires: 10 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 103.249.83.162:1369 (NAT) [Jul 16 16:37:13] NOTICE[30165]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from '"109" ;tag=F83AF917-267B271D' <--- Transmitting (NAT) to 103.249.83.162:1369 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 103.249.83.162:1369;branch=z9hG4bKec24a1a7B072E84D;received=103.249.83.162;rport=1369 From: "109" ;tag=F83AF917-267B271D To: ;tag=as6ccb33a2 Call-ID: d901a7a7-bfa24a4d-6dc3158f@192.168.4.128 CSeq: 705 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cf96805", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'd901a7a7-bfa24a4d-6dc3158f@192.168.4.128' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:1369 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 Via: SIP/2.0/UDP 103.249.83.162:1369;branch=z9hG4bKe1d27f8fCA0110C5 From: "109" ;tag=F83AF917-267B271D To: CSeq: 706 REGISTER Call-ID: d901a7a7-bfa24a4d-6dc3158f@192.168.4.128 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Authorization: Digest username="109", realm="asterisk", nonce="6cf96805", uri="sip:bmw.pulsework360.com", response="e19b6cdec7a13aed44f75279e2d8c40f", algorithm=MD5 Max-Forwards: 70 Expires: 10 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 103.249.83.162:1369 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:1369: OPTIONS sip:109@103.249.83.162:1369 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK425dc142;rport Max-Forwards: 70 From: "asterisk" ;tag=as7aaebdb3 To: Contact: Call-ID: 2f55fff81ef1ec6d6f4267d43fdbd15a@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:1369 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:1369;branch=z9hG4bKe1d27f8fCA0110C5;received=103.249.83.162;rport=1369 From: "109" ;tag=F83AF917-267B271D To: ;tag=as6ccb33a2 Call-ID: d901a7a7-bfa24a4d-6dc3158f@192.168.4.128 CSeq: 706 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:13 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '49795c1d7b342df8588005e671fc60c1@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:1369: NOTIFY sip:109@103.249.83.162:1369 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK16cc9c9f;rport Max-Forwards: 70 From: "asterisk" ;tag=as16ac031d To: Contact: Call-ID: 49795c1d7b342df8588005e671fc60c1@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog 'd901a7a7-bfa24a4d-6dc3158f@192.168.4.128' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:1369 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK425dc142;rport From: "asterisk" ;tag=as7aaebdb3 To: "109" ;tag=89B4207D-5FE46A9F CSeq: 102 OPTIONS Call-ID: 2f55fff81ef1ec6d6f4267d43fdbd15a@13.234.154.122:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '2f55fff81ef1ec6d6f4267d43fdbd15a@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:1369 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK16cc9c9f;rport From: "asterisk" ;tag=as16ac031d To: "109" ;tag=C7007BC7-FC387DAD CSeq: 102 NOTIFY Call-ID: 49795c1d7b342df8588005e671fc60c1@13.234.154.122:5060 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '49795c1d7b342df8588005e671fc60c1@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:13.234.154.122 SIP/2.0 From: ;tag=5f31e8-b404a8c0-13c4-5d29b70b-47e56b88-5d29b70b To: Call-ID: 616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b CSeq: 17338 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db0f5-fa7b406c-7d44a9ec Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="106",realm="asterisk",nonce="481afb03",uri="sip:13.234.154.122",response="333b847955731683952cf94efdb3cff0",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) [Jul 16 16:37:13] NOTICE[30165]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from ';tag=5f31e8-b404a8c0-13c4-5d29b70b-47e56b88-5d29b70b' <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db0f5-fa7b406c-7d44a9ec;received=103.249.83.162;rport=5060 From: ;tag=5f31e8-b404a8c0-13c4-5d29b70b-47e56b88-5d29b70b To: ;tag=as6c29d936 Call-ID: 616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b CSeq: 17338 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="423abae8", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:13.234.154.122 SIP/2.0 From: ;tag=5f31e8-b404a8c0-13c4-5d29b70b-47e56b88-5d29b70b To: Call-ID: 616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b CSeq: 17339 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db0f5-fa7b4097-27992a4a Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="106",realm="asterisk",nonce="423abae8",uri="sip:13.234.154.122",response="731d7ba748e59d52b28c22f249619c97",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:5060: OPTIONS sip:106@103.249.83.162:5060;transport=udp;rinstance=cjWfvSBtRH699DM SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK0c42b405;rport Max-Forwards: 70 From: "asterisk" ;tag=as78f08345 To: Contact: Call-ID: 657476991253f1911af437f508ff67f7@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db0f5-fa7b4097-27992a4a;received=103.249.83.162;rport=5060 From: ;tag=5f31e8-b404a8c0-13c4-5d29b70b-47e56b88-5d29b70b To: ;tag=as6c29d936 Call-ID: 616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b CSeq: 17339 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:13 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '34dfccae3a1db3b636d10b10762de7b6@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:5060: NOTIFY sip:106@103.249.83.162:5060;transport=udp;rinstance=cjWfvSBtRH699DM SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK488e4678;rport Max-Forwards: 70 From: "asterisk" ;tag=as6268a597 To: Contact: Call-ID: 34dfccae3a1db3b636d10b10762de7b6@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d29b70b-68fddc42-5d29b70b' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as78f08345 To: ;tag=5e8348-b404a8c0-13c4-5d2db0f5-49185288-5d2db0f5 Call-ID: 657476991253f1911af437f508ff67f7@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Matrix-SetuVFXTH Supported: replaces Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK0c42b405 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '657476991253f1911af437f508ff67f7@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as6268a597 To: ;tag=5f0f88-b404a8c0-13c4-5d2db0f5-2cf4c0a6-5d2db0f5 Call-ID: 34dfccae3a1db3b636d10b10762de7b6@13.234.154.122:5060 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK488e4678 Supported: replaces Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '34dfccae3a1db3b636d10b10762de7b6@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:1370 ---> INVITE sip:1164@13.234.154.122 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 1 INVITE Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafb9-fa766b4d-5f315571 P-Asserted-Identity: Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 278 v=0 o=- 3220474038 3220474038 IN IP4 103.249.83.162 s=Matrix-SETUVTEP c=IN IP4 103.249.83.162 t=0 0 m=audio 8016 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=rtcp:8017 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 103.249.83.162:1370 (NAT) Sending to 103.249.83.162:1370 (NAT) Using INVITE request as basis request - dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 Found peer '9' for '08807486181' from 103.249.83.162:1370 == Using SIP RTP CoS mark 5 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 - (alaw|ulaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.162:8016 Looking for 1164 in kunamt1 (domain 13.234.154.122) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafb9-fa766b4d-5f315571;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 1 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [1164@kunamt1:1] MixMonitor("SIP/9-0000003f", "/var/www/html/monitor/from-internal/1563275235.96.wav,") in new stack -- Executing [1164@kunamt1:2] Set("SIP/9-0000003f", "__src=9") in new stack -- Executing [1164@kunamt1:3] NoOp("SIP/9-0000003f", "1164 SIP/9-0000003f--9--0 ") in new stack -- Executing [1164@kunamt1:4] Set("SIP/9-0000003f", "__uniqueid=1563275235.96") in new stack -- Executing [1164@kunamt1:5] Queue("SIP/9-0000003f", "testing,trn,,,60,,qm-testing") in new stack <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafb9-fa766b4d-5f315571;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 1 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> == Begin MixMonitor Recording SIP/9-0000003f == Using SIP RTP CoS mark 5 Audio is at 49082 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 103.249.83.166:5060: INVITE sip:07904022395@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK6687627a;rport Max-Forwards: 70 From: ;tag=as7837818e To: Contact: Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 279 v=0 o=root 1095814010 1095814010 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 49082 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/07904022395@matrix == Using SIP RTP CoS mark 5 Audio is at 63372 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 103.249.83.166:5060: INVITE sip:07550197322@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK562eb10e;rport Max-Forwards: 70 From: ;tag=as5e5dd9b8 To: Contact: Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 279 v=0 o=root 1664025878 1664025878 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 63372 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/07550197322@matrix <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as7837818e To: Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK6687627a Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as5e5dd9b8 To: Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK562eb10e Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Reliably Transmitting (NAT) to 43.254.109.83:1129: OPTIONS sip:803@43.254.109.83:1129 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK23ac366c;rport Max-Forwards: 70 From: "asterisk" ;tag=as604b424e To: Contact: Call-ID: 4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 43.254.109.83:1130: OPTIONS sip:802@43.254.109.83:1130 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK17135670;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1070cf To: Contact: Call-ID: 4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 43.254.109.83:1129: OPTIONS sip:803@43.254.109.83:1129 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK23ac366c;rport Max-Forwards: 70 From: "asterisk" ;tag=as604b424e To: Contact: Call-ID: 4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 43.254.109.83:1130: OPTIONS sip:802@43.254.109.83:1130 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK17135670;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1070cf To: Contact: Call-ID: 4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 183 Session Progress From: ;tag=as5e5dd9b8 To: ;tag=da3588-af04a8c0-13c4-5d2dafbb-30845b82-5d2dafbb Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK562eb10e Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 254 v=0 o=- 3883603545 3883603545 IN IP4 103.249.83.166 s=Matrix-SETUVTEP c=IN IP4 103.249.83.166 t=0 0 m=audio 8020 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:8021 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (12 headers 11 lines) --- sip_route_dump: route/path hop: Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.166:8020 <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 183 Session Progress From: ;tag=as7837818e To: ;tag=da3428-af04a8c0-13c4-5d2dafbb-65abd1ac-5d2dafbb Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK6687627a Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 250 v=0 o=- 45493774 45493774 IN IP4 103.249.83.166 s=Matrix-SETUVTEP c=IN IP4 103.249.83.166 t=0 0 m=audio 8018 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:8019 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (12 headers 11 lines) --- sip_route_dump: route/path hop: Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.166:8018 Retransmitting #2 (NAT) to 43.254.109.83:1129: OPTIONS sip:803@43.254.109.83:1129 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK23ac366c;rport Max-Forwards: 70 From: "asterisk" ;tag=as604b424e To: Contact: Call-ID: 4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 43.254.109.83:1130: OPTIONS sip:802@43.254.109.83:1130 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK17135670;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1070cf To: Contact: Call-ID: 4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 43.254.109.83:1129: OPTIONS sip:803@43.254.109.83:1129 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK23ac366c;rport Max-Forwards: 70 From: "asterisk" ;tag=as604b424e To: Contact: Call-ID: 4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 43.254.109.83:1130: OPTIONS sip:802@43.254.109.83:1130 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK17135670;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1070cf To: Contact: Call-ID: 4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 43.254.109.83:1129: OPTIONS sip:803@43.254.109.83:1129 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK23ac366c;rport Max-Forwards: 70 From: "asterisk" ;tag=as604b424e To: Contact: Call-ID: 4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '4027358102b2dd874d0e0efd630ffd5e@13.234.154.122:5060' Method: OPTIONS Retransmitting #4 (NAT) to 43.254.109.83:1130: OPTIONS sip:802@43.254.109.83:1130 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK17135670;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e1070cf To: Contact: Call-ID: 4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '4bf6bcc108a03df1625ea186084815fa@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 200 OK From: ;tag=as7837818e To: ;tag=da3428-af04a8c0-13c4-5d2dafbb-65abd1ac-5d2dafbb Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK6687627a Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 250 v=0 o=- 45493774 45493774 IN IP4 103.249.83.166 s=Matrix-SETUVTEP c=IN IP4 103.249.83.166 t=0 0 m=audio 8018 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:8019 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (12 headers 11 lines) --- sip_route_dump: route/path hop: Transmitting (NAT) to 103.249.83.166:5060: ACK sip:103.249.83.166:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK006e29d4;rport Max-Forwards: 70 From: ;tag=as7837818e To: ;tag=da3428-af04a8c0-13c4-5d2dafbb-65abd1ac-5d2dafbb Contact: Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.2.1 Content-Length: 0 --- -- SIP/matrix-00000040 answered SIP/9-0000003f Scheduling destruction of SIP dialog '52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 103.249.83.166:5060: CANCEL sip:07550197322@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK562eb10e;rport Max-Forwards: 70 From: ;tag=as5e5dd9b8 To: Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 16.2.1 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 --- Scheduling destruction of SIP dialog '52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060' in 6400 ms (Method: INVITE) -- Executing [s@macro-qm-testing:1] NoOp("SIP/matrix-00000040", "SIP/07904022395@matrix --SIP/07904022395@matrix ") in new stack -- Executing [s@macro-qm-testing:2] MYSQL("SIP/matrix-00000040", "Connect connid 127.0.0.1 root Pulse@123 asterisk") in new stack -- Executing [s@macro-qm-testing:3] GotoIf("SIP/matrix-00000040", "0?error,1") in new stack -- Executing [s@macro-qm-testing:4] MYSQL("SIP/matrix-00000040", "Query resultid 2 insert into Group_dst (Dst,Uniqueid)values("SIP/07904022395@matrix",1563275235.96)") in new stack <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 200 OK From: ;tag=as5e5dd9b8 To: ;tag=da3588-af04a8c0-13c4-5d2dafbb-30845b82-5d2dafbb Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 CANCEL Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK562eb10e Supported: replaces User-Agent: Matrix-SETUVTEP Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 487 Request Terminated From: ;tag=as5e5dd9b8 To: ;tag=da3588-af04a8c0-13c4-5d2dafbb-30845b82-5d2dafbb Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK562eb10e Supported: replaces User-Agent: Matrix-SETUVTEP Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 103.249.83.166:5060: ACK sip:103.249.83.166:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK562eb10e;rport Max-Forwards: 70 From: ;tag=as5e5dd9b8 To: ;tag=da3588-af04a8c0-13c4-5d2dafbb-30845b82-5d2dafbb Contact: Call-ID: 52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.2.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060' in 6400 ms (Method: INVITE) -- Executing [s@macro-qm-testing:5] Set("SIP/matrix-00000040", "curresul= -- 0x213 - Invalid Destination numbersuccess") in new stack Audio is at 41356 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafb9-fa766b4d-5f315571;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 1 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 277 v=0 o=root 766810138 766810138 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 41356 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/matrix-00000040 joined 'simple_bridge' basic-bridge <63234c60-1a92-4a8f-8f51-f24d9b75b086> -- Channel SIP/9-0000003f joined 'simple_bridge' basic-bridge <63234c60-1a92-4a8f-8f51-f24d9b75b086> <--- SIP read from UDP:103.249.83.162:1370 ---> ACK sip:1164@13.234.154.122:5060 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 1 ACK Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafbe-fa7681c9-711c4a89 Max-Forwards: 70 User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.162:1370 ---> INVITE sip:1164@13.234.154.122:5060 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 INVITE Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafbe-fa7681cf-249f73f Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 254 v=0 o=- 3220474038 3220474039 IN IP4 103.249.83.162 s=Matrix-SETUVTEP c=IN IP4 103.249.83.162 t=0 0 m=audio 8016 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:8017 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (13 headers 11 lines) --- Sending to 103.249.83.162:1370 (NAT) Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.162:8016 <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafbe-fa7681cf-249f73f;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 41356 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafbe-fa7681cf-249f73f;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 253 v=0 o=root 766810138 766810139 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 41356 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> Retransmitting #1 (NAT) to 103.249.83.162:1370: SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafbe-fa7681cf-249f73f;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 253 v=0 o=root 766810138 766810139 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 41356 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:103.249.83.162:1370 ---> ACK sip:1164@13.234.154.122:5060 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 ACK Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafbf-fa76824c-7da0e42f Max-Forwards: 70 User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.162:1370 ---> ACK sip:1164@13.234.154.122:5060 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 2 ACK Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafbf-fa76824c-7da0e42f Max-Forwards: 70 User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.162:1371 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 Via: SIP/2.0/UDP 103.249.83.162:1371;branch=z9hG4bKc1f75c1f78E72A09 From: "Recption" ;tag=72497D0F-40D3DD59 To: CSeq: 69 REGISTER Call-ID: bc0d0ff-801611a9-7b485757@192.168.4.127 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Authorization: Digest username="9", realm="asterisk", nonce="2b47eb39", uri="sip:bmw.pulsework360.com", response="6ecba883051c7b99b2146fa07c72bf9a", algorithm=MD5 Max-Forwards: 70 Expires: 10 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 103.249.83.162:1371 (NAT) [Jul 16 16:37:22] NOTICE[30165]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from '"Recption" ;tag=72497D0F-40D3DD59' <--- Transmitting (NAT) to 103.249.83.162:1371 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 103.249.83.162:1371;branch=z9hG4bKc1f75c1f78E72A09;received=103.249.83.162;rport=1371 From: "Recption" ;tag=72497D0F-40D3DD59 To: ;tag=as09ad8771 Call-ID: bc0d0ff-801611a9-7b485757@192.168.4.127 CSeq: 69 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75143a34", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'bc0d0ff-801611a9-7b485757@192.168.4.127' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:1371 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 Via: SIP/2.0/UDP 103.249.83.162:1371;branch=z9hG4bK9b51cd77F912CF51 From: "Recption" ;tag=72497D0F-40D3DD59 To: CSeq: 70 REGISTER Call-ID: bc0d0ff-801611a9-7b485757@192.168.4.127 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Authorization: Digest username="9", realm="asterisk", nonce="75143a34", uri="sip:bmw.pulsework360.com", response="03cfc02e00fe8664473117d36a19d48c", algorithm=MD5 Max-Forwards: 70 Expires: 10 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 103.249.83.162:1371 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:1371: OPTIONS sip:9@103.249.83.162:1371 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK38cb0387;rport Max-Forwards: 70 From: "asterisk" ;tag=as3abb5b93 To: Contact: Call-ID: 5733d9312a74d4e77350a43248117ecf@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:1371 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:1371;branch=z9hG4bK9b51cd77F912CF51;received=103.249.83.162;rport=1371 From: "Recption" ;tag=72497D0F-40D3DD59 To: ;tag=as09ad8771 Call-ID: bc0d0ff-801611a9-7b485757@192.168.4.127 CSeq: 70 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:22 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '34d55d6f4db406a76940db2431980e04@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:1371: NOTIFY sip:9@103.249.83.162:1371 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK7afadec9;rport Max-Forwards: 70 From: "asterisk" ;tag=as47669780 To: Contact: Call-ID: 34d55d6f4db406a76940db2431980e04@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog 'bc0d0ff-801611a9-7b485757@192.168.4.127' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:1371 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK38cb0387;rport From: "asterisk" ;tag=as3abb5b93 To: "Recption" ;tag=FBB4EB59-E2C74EE7 CSeq: 102 OPTIONS Call-ID: 5733d9312a74d4e77350a43248117ecf@13.234.154.122:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '5733d9312a74d4e77350a43248117ecf@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:1371 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK7afadec9;rport From: "asterisk" ;tag=as47669780 To: "Recption" ;tag=58FDFAFF-E9AD3FA9 CSeq: 102 NOTIFY Call-ID: 34d55d6f4db406a76940db2431980e04@13.234.154.122:5060 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '34d55d6f4db406a76940db2431980e04@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 From: ;tag=5ea9c8-b404a8c0-13c4-5d2d64a8-169cc913-5d2d64a8 To: Call-ID: 616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8 CSeq: 1322 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db0ff-fa7b677c-21ecf2a4 Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="133",realm="asterisk",nonce="70175845",uri="sip:bmw.pulsework360.com",response="78cb299c152b3c398b1b1226a57431ec",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) [Jul 16 16:37:23] NOTICE[30165]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from ';tag=5ea9c8-b404a8c0-13c4-5d2d64a8-169cc913-5d2d64a8' <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db0ff-fa7b677c-21ecf2a4;received=103.249.83.162;rport=5060 From: ;tag=5ea9c8-b404a8c0-13c4-5d2d64a8-169cc913-5d2d64a8 To: ;tag=as32a8ebae Call-ID: 616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8 CSeq: 1322 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a79dc73", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 From: ;tag=5ea9c8-b404a8c0-13c4-5d2d64a8-169cc913-5d2d64a8 To: Call-ID: 616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8 CSeq: 1323 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db0ff-fa7b67a7-57294682 Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="133",realm="asterisk",nonce="5a79dc73",uri="sip:bmw.pulsework360.com",response="5bfe5783d90b52514ded2953a80f8a14",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:5060: OPTIONS sip:133@103.249.83.162:5060;transport=udp;rinstance=L7DpYzxny1PXxDu SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK2f98beb4;rport Max-Forwards: 70 From: "asterisk" ;tag=as111b5777 To: Contact: Call-ID: 511933e9036040e653b15a47052aa472@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db0ff-fa7b67a7-57294682;received=103.249.83.162;rport=5060 From: ;tag=5ea9c8-b404a8c0-13c4-5d2d64a8-169cc913-5d2d64a8 To: ;tag=as32a8ebae Call-ID: 616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8 CSeq: 1323 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:23 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4e96d63e08ff6098794783e639be3ae3@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:5060: NOTIFY sip:133@103.249.83.162:5060;transport=udp;rinstance=L7DpYzxny1PXxDu SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK30b84302;rport Max-Forwards: 70 From: "asterisk" ;tag=as20de965b To: Contact: Call-ID: 4e96d63e08ff6098794783e639be3ae3@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d2d64a8-780fe71d-5d2d64a8' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as111b5777 To: ;tag=5e6be8-b404a8c0-13c4-5d2db0ff-28f9fa40-5d2db0ff Call-ID: 511933e9036040e653b15a47052aa472@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Matrix-SetuVFXTH Supported: replaces Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK2f98beb4 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '511933e9036040e653b15a47052aa472@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as20de965b To: ;tag=5ee4e8-b404a8c0-13c4-5d2db0ff-7b752bde-5d2db0ff Call-ID: 4e96d63e08ff6098794783e639be3ae3@13.234.154.122:5060 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK30b84302 Supported: replaces Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '4e96d63e08ff6098794783e639be3ae3@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:103.249.83.162:1370 ---> BYE sip:1164@13.234.154.122:5060 SIP/2.0 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 3 BYE Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafc1-fa768d9f-163245bd Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SETUVTEP Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 103.249.83.162:1370 (NAT) Scheduling destruction of SIP dialog 'dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafc1-fa768d9f-163245bd;received=103.249.83.162;rport=1370 From: ;tag=da44a8-af04a8c0-13c4-5d2dafb9-11842f3b-5d2dafb9 To: ;tag=as313944fb Call-ID: dc1c78-af04a8c0-13c4-5d2dafb9-680291cd-5d2dafb9 CSeq: 3 BYE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/9-0000003f left 'simple_bridge' basic-bridge <63234c60-1a92-4a8f-8f51-f24d9b75b086> == Spawn extension (kunamt1, 1164, 5) exited non-zero on 'SIP/9-0000003f' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/9-0000003f -- Channel SIP/matrix-00000040 left 'simple_bridge' basic-bridge <63234c60-1a92-4a8f-8f51-f24d9b75b086> Scheduling destruction of SIP dialog '4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 103.249.83.166:5060: BYE sip:103.249.83.166:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK53f8458c;rport Max-Forwards: 70 From: ;tag=as7837818e To: ;tag=da3428-af04a8c0-13c4-5d2dafbb-65abd1ac-5d2dafbb Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 16.2.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 200 OK From: ;tag=as7837818e To: ;tag=da3428-af04a8c0-13c4-5d2dafbb-65abd1ac-5d2dafbb Call-ID: 4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060 CSeq: 103 BYE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK53f8458c Supported: replaces User-Agent: Matrix-SETUVTEP Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4f02f03e436265ef72d3d4a7041535e4@13.234.154.122:5060' Method: INVITE <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 From: ;tag=5e08e8-b404a8c0-13c4-5d29b709-16c6aedd-5d29b709 To: Call-ID: 616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709 CSeq: 17326 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db100-fa7b6b6e-20f4d95c Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="104",realm="asterisk",nonce="51af9413",uri="sip:bmw.pulsework360.com",response="249f5d3ecbf6c6c4be8d661bcdeaf002",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) [Jul 16 16:37:24] NOTICE[30165]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from ';tag=5e08e8-b404a8c0-13c4-5d29b709-16c6aedd-5d29b709' <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db100-fa7b6b6e-20f4d95c;received=103.249.83.162;rport=5060 From: ;tag=5e08e8-b404a8c0-13c4-5d29b709-16c6aedd-5d29b709 To: ;tag=as298799d9 Call-ID: 616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709 CSeq: 17326 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30c52f7a", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> REGISTER sip:bmw.pulsework360.com SIP/2.0 From: ;tag=5e08e8-b404a8c0-13c4-5d29b709-16c6aedd-5d29b709 To: Call-ID: 616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709 CSeq: 17327 REGISTER Via: SIP/2.0/UDP 103.249.83.162:5060;rport;branch=z9hG4bK-5d2db100-fa7b6b98-2a74e0ba Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SetuVFXTH Contact: Expires: 60 Authorization: Digest username="104",realm="asterisk",nonce="30c52f7a",uri="sip:bmw.pulsework360.com",response="44dabe3330a2b3b85eefacd6767da148",algorithm=MD5 Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 103.249.83.162:5060 (NAT) Reliably Transmitting (NAT) to 103.249.83.162:5060: OPTIONS sip:104@103.249.83.162:5060;transport=udp;rinstance=t0Ikq9uV3B4ujki SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK47aa61ee;rport Max-Forwards: 70 From: "asterisk" ;tag=as29ff519a To: Contact: Call-ID: 22b1aec13f7e2149297ee9330189e7de@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (NAT) to 103.249.83.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 103.249.83.162:5060;branch=z9hG4bK-5d2db100-fa7b6b98-2a74e0ba;received=103.249.83.162;rport=5060 From: ;tag=5e08e8-b404a8c0-13c4-5d29b709-16c6aedd-5d29b709 To: ;tag=as298799d9 Call-ID: 616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709 CSeq: 17327 REGISTER Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 16 Jul 2019 11:07:24 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2a3da7d638d419503e3e33ef1a9b3c26@13.234.154.122:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 103.249.83.162:5060: NOTIFY sip:104@103.249.83.162:5060;transport=udp;rinstance=t0Ikq9uV3B4ujki SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK01616f70;rport Max-Forwards: 70 From: "asterisk" ;tag=as305676ea To: Contact: Call-ID: 2a3da7d638d419503e3e33ef1a9b3c26@13.234.154.122:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.2.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@13.234.154.122 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '616660-b404a8c0-13c4-5d29b709-35c22447-5d29b709' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as29ff519a To: ;tag=5efda8-b404a8c0-13c4-5d2db100-5fd0fff8-5d2db100 Call-ID: 22b1aec13f7e2149297ee9330189e7de@13.234.154.122:5060 CSeq: 102 OPTIONS User-Agent: Matrix-SetuVFXTH Supported: replaces Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK47aa61ee Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '22b1aec13f7e2149297ee9330189e7de@13.234.154.122:5060' Method: OPTIONS <--- SIP read from UDP:103.249.83.162:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as305676ea To: ;tag=5ed728-b404a8c0-13c4-5d2db100-7de2d516-5d2db100 Call-ID: 2a3da7d638d419503e3e33ef1a9b3c26@13.234.154.122:5060 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK01616f70 Supported: replaces Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2a3da7d638d419503e3e33ef1a9b3c26@13.234.154.122:5060' Method: NOTIFY <--- SIP read from UDP:43.254.109.83:5060 ---> <-------------> <--- SIP read from UDP:103.249.83.162:1370 ---> INVITE sip:1157@13.234.154.122 SIP/2.0 From: ;tag=da4608-af04a8c0-13c4-5d2dafc3-1fdffbd7-5d2dafc3 To: Call-ID: dc2f98-af04a8c0-13c4-5d2dafc3-451ab5a1-5d2dafc3 CSeq: 1 INVITE Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2dafc3-fa76931b-722a2314 P-Asserted-Identity: Max-Forwards: 70 Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 278 v=0 o=- 1674084270 1674084270 IN IP4 103.249.83.162 s=Matrix-SETUVTEP c=IN IP4 103.249.83.162 t=0 0 m=audio 8022 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=rtcp:8023 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 103.249.83.162:1370 (NAT) Sending to 103.249.83.162:1370 (NAT) Using INVITE request as basis request - dc2f98-af04a8c0-13c4-5d2dafc3-451ab5a1-5d2dafc3 Found peer '111' for '09894963116' from 103.249.83.162:1370 == Using SIP RTP CoS mark 5 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 - (alaw|ulaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.162:8022 Looking for 1157 in kunamt1 (domain 13.234.154.122) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafc3-fa76931b-722a2314;received=103.249.83.162;rport=1370 From: ;tag=da4608-af04a8c0-13c4-5d2dafc3-1fdffbd7-5d2dafc3 To: Call-ID: dc2f98-af04a8c0-13c4-5d2dafc3-451ab5a1-5d2dafc3 CSeq: 1 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:43.254.109.83:1153 ---> <-------------> -- Executing [1157@kunamt1:1] MixMonitor("SIP/111-00000042", "/var/www/html/monitor/from-internal/1563275245.101.wav,") in new stack -- Executing [1157@kunamt1:2] Set("SIP/111-00000042", "__src=111") in new stack -- Executing [1157@kunamt1:3] Set("SIP/111-00000042", "__uniqueid=1563275245.101") in new stack -- Executing [1157@kunamt1:4] Queue("SIP/111-00000042", "coimbatoread,trn,,,60,,qm-coimbatoread") in new stack <--- Transmitting (NAT) to 103.249.83.162:1370 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2dafc3-fa76931b-722a2314;received=103.249.83.162;rport=1370 From: ;tag=da4608-af04a8c0-13c4-5d2dafc3-1fdffbd7-5d2dafc3 To: ;tag=as696981dc Call-ID: dc2f98-af04a8c0-13c4-5d2dafc3-451ab5a1-5d2dafc3 CSeq: 1 INVITE Server: Asterisk PBX 16.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> == Begin MixMonitor Recording SIP/111-00000042 == Using SIP RTP CoS mark 5 Audio is at 5744 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 103.249.83.166:5060: INVITE sip:09094484966@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK5fe3582d;rport Max-Forwards: 70 From: ;tag=as05047d62 To: Contact: Call-ID: 49f36b0c13bea9eb45d8fe51027d3bb7@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 278 v=0 o=root 2144037954 2144037954 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 5744 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/09094484966@matrix == Using SIP RTP CoS mark 5 Audio is at 57340 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 103.249.83.166:5060: INVITE sip:09941167711@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK44a9e9e7;rport Max-Forwards: 70 From: ;tag=as1d91db2c To: Contact: Call-ID: 743ab9d75367b8fb4a43545a63710483@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 279 v=0 o=root 2132610474 2132610474 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 57340 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/09941167711@matrix Retransmitting #1 (NAT) to 103.249.83.166:5060: INVITE sip:09094484966@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK5fe3582d;rport Max-Forwards: 70 From: ;tag=as05047d62 To: Contact: Call-ID: 49f36b0c13bea9eb45d8fe51027d3bb7@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 278 v=0 o=root 2144037954 2144037954 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 5744 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as05047d62 To: Call-ID: 49f36b0c13bea9eb45d8fe51027d3bb7@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK5fe3582d Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Retransmitting #1 (NAT) to 103.249.83.166:5060: INVITE sip:09941167711@103.249.83.166 SIP/2.0 Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK44a9e9e7;rport Max-Forwards: 70 From: ;tag=as1d91db2c To: Contact: Call-ID: 743ab9d75367b8fb4a43545a63710483@13.234.154.122:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.2.1 Date: Tue, 16 Jul 2019 11:07:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 279 v=0 o=root 2132610474 2132610474 IN IP4 13.234.154.122 s=Asterisk PBX 16.2.1 c=IN IP4 13.234.154.122 t=0 0 m=audio 57340 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as1d91db2c To: Call-ID: 743ab9d75367b8fb4a43545a63710483@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK44a9e9e7 Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as05047d62 To: Call-ID: 49f36b0c13bea9eb45d8fe51027d3bb7@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK5fe3582d Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 100 Trying From: ;tag=as1d91db2c To: Call-ID: 743ab9d75367b8fb4a43545a63710483@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK44a9e9e7 Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:43.254.109.83:5061 ---> <-------------> <--- SIP read from UDP:43.254.109.83:1154 ---> <-------------> Really destroying SIP dialog '52866b513fe8670b0acaf19d32e75ef5@13.234.154.122:5060' Method: INVITE <--- SIP read from UDP:103.249.83.166:5060 ---> SIP/2.0 183 Session Progress From: ;tag=as05047d62 To: ;tag=da5c08-af04a8c0-13c4-5d2dafc5-11645242-5d2dafc5 Call-ID: 49f36b0c13bea9eb45d8fe51027d3bb7@13.234.154.122:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 13.234.154.122:5060;rport=5060;branch=z9hG4bK5fe3582d Supported: replaces User-Agent: Matrix-SETUVTEP Contact: Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS Content-Type: application/sdp Content-Length: 254 v=0 o=- 3390043817 3390043817 IN IP4 103.249.83.166 s=Matrix-SETUVTEP c=IN IP4 103.249.83.166 t=0 0 m=audio 8024 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:8025 IN IP4 103.249.83.166 a=sendrecv <-------------> --- (12 headers 11 lines) --- sip_route_dump: route/path hop: Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 103.249.83.166:8024 ip-10-0-1-103*CLI> sip set debug off SIP Debugging Disabled -- SIP/matrix-0000003e answered SIP/116-0000003d -- Executing [s@macro-qm-accessories:1] NoOp("SIP/matrix-0000003e", "SIP/09940398942@matrix --SIP/09940398942@matrix ") in new stack -- Executing [s@macro-qm-accessories:2] MYSQL("SIP/matrix-0000003e", "Connect connid 127.0.0.1 root Pulse@123 asterisk") in new stack -- Executing [s@macro-qm-accessories:3] GotoIf("SIP/matrix-0000003e", "0?error,1") in new stack -- Executing [s@macro-qm-accessories:4] MYSQL("SIP/matrix-0000003e", "Query resultid 2 insert into Group_dst (Dst,Uniqueid)values("SIP/09940398942@matrix",1563275228.94)") in new stack -- Executing [s@macro-qm-accessories:5] Set("SIP/matrix-0000003e", "curresul= -- 0x213 - Invalid Destination numbersuccess") in new stack -- Channel SIP/matrix-0000003e joined 'simple_bridge' basic-bridge <71706163-7e4b-4434-aed3-3bb21cdd9fb2> -- Channel SIP/116-0000003d joined 'simple_bridge' basic-bridge <71706163-7e4b-4434-aed3-3bb21cdd9fb2> ip-10-0-1-103*CLI> exit Asterisk cleanly ending (0). Executing last minute cleanups [root@ip-10-0-1-103 centos]#