<-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> INVITE sip:14134425424@aron.com:5060 SIP/2.0 Call-ID: 179d9786@192.168.1.107 Content-Length: 314 CSeq: 8001 INVITE From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-42665f0e;rport User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: Expires: 60 Supported: replaces Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Remote-Party-ID: ;party=calling;privacy=off Content-Type: application/sdp v=0 o=- 35001853 1 IN IP4 67.86.108.167 s=- c=IN IP4 67.86.108.167 t=0 0 m=audio 16664 RTP/AVP 0 8 18 104 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:104 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 a=xg726bitorder:big-endian <-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: --- (15 headers 15 lines) --- [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Using INVITE request as basis request - 179d9786@192.168.1.107 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found peer '8001' for '8001' from 67.86.108.167:5060 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: <--- Reliably Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-42665f0e;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as5865faf0 Call-ID: 179d9786@192.168.1.107 CSeq: 8001 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="13e21a9a" Content-Length: 0 <------------> [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Scheduling destruction of SIP dialog '179d9786@192.168.1.107' in 32000 ms (Method: INVITE) [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> ACK sip:14134425424@aron.com:5060 SIP/2.0 Call-ID: 179d9786@192.168.1.107 Content-Length: 0 CSeq: 8001 ACK From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: ;tag=as5865faf0 Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-42665f0e;rport User-Agent: OBIHAI/OBi110-1.3.0.2860 <-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: --- (9 headers 0 lines) --- [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> INVITE sip:14134425424@aron.com:5060 SIP/2.0 Call-ID: 179d9786@192.168.1.107 Content-Length: 314 CSeq: 8002 INVITE From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;rport Authorization: DIGEST algorithm=MD5,nonce="13e21a9a",realm="asterisk",response="24972cbc25852cce4a50fb9610cfc3a3",uri="sip:14134425424@aron.com:5060",username="8001" User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: Expires: 60 Supported: replaces Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Remote-Party-ID: ;party=calling;privacy=off Content-Type: application/sdp v=0 o=- 35001853 1 IN IP4 67.86.108.167 s=- c=IN IP4 67.86.108.167 t=0 0 m=audio 16664 RTP/AVP 0 8 18 104 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:104 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 a=xg726bitorder:big-endian <-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: --- (16 headers 15 lines) --- [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Using INVITE request as basis request - 179d9786@192.168.1.107 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found peer '8001' for '8001' from 67.86.108.167:5060 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 0 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 8 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 18 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 104 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 101 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format G729 for ID 18 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format G726-32 for ID 104 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g729|g726)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Peer audio RTP is at port 67.86.108.167:16664 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Looking for 14134425424 in twerski (domain aron.com) [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] sip/route.c: sip_route_dump: route/path hop: [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 7 23:08:32] VERBOSE[20478][C-0000032d] chan_sip.c: Audio is at 14508 [Mar 7 23:08:32] VERBOSE[20478][C-0000032d] chan_sip.c: Adding codec ulaw to SDP [Mar 7 23:08:32] VERBOSE[20478][C-0000032d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 7 23:08:32] VERBOSE[20478][C-0000032d] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: INVITE sip:14134425424@sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK7ae96d65;rport Max-Forwards: 70 From: ;tag=as49be75cb To: Contact: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Date: Tue, 08 Mar 2022 04:08:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8001" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 253 v=0 o=root 537701804 537701804 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 14508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:32] VERBOSE[20478][C-0000032d] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK7ae96d65;rport=5060 From: ;tag=as49be75cb To: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 102 INVITE <-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: --- (6 headers 0 lines) --- [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK7ae96d65;rport=5060 From: ;tag=as49be75cb To: ;tag=5ed9.5caa050f64bca8649fb1a23819794607 Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="192.168.1.1", nonce="6226d6de00007299fc7762e2bd4e9d6fd7b0e5a47d6a6872" Content-Length: 0 <-------------> [Mar 7 23:08:32] VERBOSE[31926] chan_sip.c: --- (8 headers 0 lines) --- [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Transmitting (NAT) to 76.8.29.198:5060: ACK sip:14134425424@sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK7ae96d65;rport Max-Forwards: 70 From: ;tag=as49be75cb To: ;tag=5ed9.5caa050f64bca8649fb1a23819794607 Contact: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Content-Length: 0 --- [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Audio is at 14508 [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Adding codec ulaw to SDP [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 7 23:08:32] VERBOSE[31926][C-0000032d] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: INVITE sip:14134425424@sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK698eae5c;rport Max-Forwards: 70 From: ;tag=as49be75cb To: Contact: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Proxy-Authorization: Digest username="609172_Aron", realm="192.168.1.1", algorithm=MD5, uri="sip:14134425424@sip.bulkvs.com", nonce="6226d6de00007299fc7762e2bd4e9d6fd7b0e5a47d6a6872", response="5f83ee645c3ee3178e5c5b841ec26c98" Date: Tue, 08 Mar 2022 04:08:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8001" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 253 v=0 o=root 537701804 537701805 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 14508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:33] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK698eae5c;rport=5060 From: ;tag=as49be75cb To: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 103 INVITE <-------------> [Mar 7 23:08:33] VERBOSE[31926] chan_sip.c: --- (6 headers 0 lines) --- [Mar 7 23:08:35] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_Aron@sip.bulkvs.com [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK383950eb;rport Max-Forwards: 70 From: ;tag=as76317b98 To: Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1775 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_Aron", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d64000006e0250140ced45eb98a9389aad9ce1a76c19", response="f688bf2cd94cf287d558da61938f4c16" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:35] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_770@sip.bulkvs.com [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4f41971f;rport Max-Forwards: 70 From: ;tag=as417b63cf To: Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1775 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_770", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d64000006e039e1c614c51c2cb48cb80a93760e9a2ea", response="6788a98a334f582b1733ef097e243c79" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK383950eb;rport=5060 From: ;tag=as76317b98 To: ;tag=5ed9.462b0b0b936bbabc903dd806c3c15abe Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1775 REGISTER WWW-Authenticate: Digest realm="sip.bulkvs.com", nonce="6226d6e1000072b77ac15235ecbe96ac995e872c1d5e7d1e", stale=true Content-Length: 0 <-------------> [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: --- (8 headers 0 lines) --- [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Responding to challenge, registration to domain/host name sip.bulkvs.com [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK6e9d46f9;rport Max-Forwards: 70 From: ;tag=as76317b98 To: Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1776 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_Aron", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072b77ac15235ecbe96ac995e872c1d5e7d1e", response="67a0546c78e4b0f752c7171177ce195f" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK4f41971f;rport=5060 From: ;tag=as417b63cf To: ;tag=5ed9.3adc21a69dbf921a553f5d6f8b49ccf0 Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1775 REGISTER WWW-Authenticate: Digest realm="sip.bulkvs.com", nonce="6226d6e1000072ba682d84a98f217fadee0842ef8bf83322", stale=true Content-Length: 0 <-------------> [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: --- (8 headers 0 lines) --- [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Responding to challenge, registration to domain/host name sip.bulkvs.com [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK724f8c18;rport Max-Forwards: 70 From: ;tag=as417b63cf To: Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1776 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_770", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072ba682d84a98f217fadee0842ef8bf83322", response="c924ff52bf866d6e7b51a02d3a9fdd1a" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK6e9d46f9;rport=5060 From: ;tag=as76317b98 To: ;tag=5ed9.1d0392bfa35d1a326733a039ef14871e Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1776 REGISTER Contact: ;expires=30 Content-Length: 0 <-------------> [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: --- (8 headers 0 lines) --- [Mar 7 23:08:35] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1' Method: REGISTER [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK724f8c18;rport=5060 From: ;tag=as417b63cf To: ;tag=5ed9.a05653b0a6be1409d1592867e0ca5e87 Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1776 REGISTER Contact: ;expires=30 Content-Length: 0 <-------------> [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: --- (8 headers 0 lines) --- [Mar 7 23:08:35] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:08:35] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1' Method: REGISTER [Mar 7 23:08:37] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK698eae5c;rport=5060 From: ;tag=as49be75cb To: ;tag=gK04831af0 Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 103 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 206 Content-Disposition: session; handling=required Content-Type: application/sdp X-Remote-IP: 208.69.83.28 X-Remote-SDP-Port: m=audio 13094 RTP/AVP 0 v=0 o=- 18464719 797233 IN IP4 76.8.29.198 s=- c=IN IP4 76.8.29.198 t=0 0 m=audio 17754 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-------------> [Mar 7 23:08:37] VERBOSE[31926] chan_sip.c: --- (13 headers 11 lines) --- [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] sip/route.c: sip_route_dump: route/path hop: [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 0 [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Found RTP audio format 101 [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 7 23:08:37] VERBOSE[31926][C-0000032d] chan_sip.c: Peer audio RTP is at port 76.8.29.198:17754 [Mar 7 23:08:42] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK698eae5c;rport=5060 From: ;tag=as49be75cb To: ;tag=gK04831af0 Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 103 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Require: timer Supported: timer Session-Expires: 3000; refresher=uas Content-Length: 206 Content-Disposition: session; handling=required Content-Type: application/sdp X-Remote-IP: 208.69.83.28 X-Remote-SDP-Port: m=audio 13094 RTP/AVP 0 v=0 o=- 18464719 797233 IN IP4 76.8.29.198 s=- c=IN IP4 76.8.29.198 t=0 0 m=audio 17754 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-------------> [Mar 7 23:08:42] VERBOSE[31926] chan_sip.c: --- (17 headers 11 lines) --- [Mar 7 23:08:42] VERBOSE[31926][C-0000032d] sip/route.c: sip_route_dump: route/path hop: [Mar 7 23:08:42] VERBOSE[31926][C-0000032d] chan_sip.c: Transmitting (NAT) to 76.8.29.198:5060: ACK sip:76.8.29.198:5060;did=e09.866cebf7;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK1272168e;rport Max-Forwards: 70 From: ;tag=as49be75cb To: ;tag=gK04831af0 Contact: Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Content-Length: 0 --- [Mar 7 23:08:42] VERBOSE[20478][C-0000032d] chan_sip.c: Audio is at 11812 [Mar 7 23:08:42] VERBOSE[20478][C-0000032d] chan_sip.c: Adding codec ulaw to SDP [Mar 7 23:08:42] VERBOSE[20478][C-0000032d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 7 23:08:42] VERBOSE[20478][C-0000032d] chan_sip.c: <--- Reliably Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Mar 7 23:08:43] VERBOSE[31926] chan_sip.c: Retransmitting #1 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:44] VERBOSE[31926] chan_sip.c: Retransmitting #2 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: Retransmitting #3 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> REGISTER sip:aron.com:5060 SIP/2.0 Call-ID: 2a91e9d1@192.168.1.107 Content-Length: 0 CSeq: 88086 REGISTER From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-7b22f7b5;rport Authorization: DIGEST algorithm=MD5,nonce="5dfe4fe4",realm="asterisk",response="b20056ab2142c744528e046c6d8c9143",uri="sip:aron.com:5060",username="8001" User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:08:46] NOTICE[31926] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=SP1f9a94953c768237' [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-7b22f7b5;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as26d49318 Call-ID: 2a91e9d1@192.168.1.107 CSeq: 88086 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e4fac1f", stale=true Content-Length: 0 <------------> [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '2a91e9d1@192.168.1.107' in 32000 ms (Method: REGISTER) [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> REGISTER sip:aron.com:5060 SIP/2.0 Call-ID: 2a91e9d1@192.168.1.107 Content-Length: 0 CSeq: 88087 REGISTER From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-5ca811ad;rport Authorization: DIGEST algorithm=MD5,nonce="2e4fac1f",realm="asterisk",response="630dfb8baaa3e4d84c80a20161ba2784",uri="sip:aron.com:5060",username="8001" User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-5ca811ad;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as26d49318 Call-ID: 2a91e9d1@192.168.1.107 CSeq: 88087 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 08 Mar 2022 04:08:46 GMT Content-Length: 0 <------------> [Mar 7 23:08:46] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '2a91e9d1@192.168.1.107' in 32000 ms (Method: REGISTER) [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:192.168.1.44:5060 ---> REGISTER sip:192.168.1.1:5060 SIP/2.0 Call-ID: 8a3ae283@192.168.2.112 Content-Length: 0 CSeq: 42214 REGISTER From: ;tag=SP15a93efec3cd4da42 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-87c11a8;rport Authorization: DIGEST algorithm=MD5,nonce="51809cc5",realm="asterisk",response="973e5e96d983522bea56c47d077f8a04",uri="sip:192.168.1.1:5060",username="8180" User-Agent: OBIHAI/OBi110-1.3.0.2886 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: Sending to 192.168.1.44:5060 (NAT) [Mar 7 23:08:47] NOTICE[31926] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=SP15a93efec3cd4da42' [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.44:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-87c11a8;received=192.168.1.44;rport=5060 From: ;tag=SP15a93efec3cd4da42 To: ;tag=as34bf175f Call-ID: 8a3ae283@192.168.2.112 CSeq: 42214 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="135d94f3", stale=true Content-Length: 0 <------------> [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '8a3ae283@192.168.2.112' in 32000 ms (Method: REGISTER) [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:192.168.1.44:5060 ---> REGISTER sip:192.168.1.1:5060 SIP/2.0 Call-ID: 8a3ae283@192.168.2.112 Content-Length: 0 CSeq: 42215 REGISTER From: ;tag=SP15a93efec3cd4da42 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-699259d8;rport Authorization: DIGEST algorithm=MD5,nonce="135d94f3",realm="asterisk",response="3e031510ef652eda2ca71fa70434e1cd",uri="sip:192.168.1.1:5060",username="8180" User-Agent: OBIHAI/OBi110-1.3.0.2886 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: Sending to 192.168.1.44:5060 (NAT) [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-699259d8;received=192.168.1.44;rport=5060 From: ;tag=SP15a93efec3cd4da42 To: ;tag=as34bf175f Call-ID: 8a3ae283@192.168.2.112 CSeq: 42215 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 08 Mar 2022 04:08:47 GMT Content-Length: 0 <------------> [Mar 7 23:08:47] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '8a3ae283@192.168.2.112' in 32000 ms (Method: REGISTER) [Mar 7 23:08:50] VERBOSE[31926] chan_sip.c: Retransmitting #4 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:54] VERBOSE[31926] chan_sip.c: Retransmitting #5 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:08:55] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_Aron@sip.bulkvs.com [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4b82b47a;rport Max-Forwards: 70 From: ;tag=as76317b98 To: Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1777 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_Aron", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072b77ac15235ecbe96ac995e872c1d5e7d1e", response="67a0546c78e4b0f752c7171177ce195f" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:55] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_770@sip.bulkvs.com [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK45c60339;rport Max-Forwards: 70 From: ;tag=as417b63cf To: Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1777 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_770", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072ba682d84a98f217fadee0842ef8bf83322", response="c924ff52bf866d6e7b51a02d3a9fdd1a" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK4b82b47a;rport=5060 From: ;tag=as76317b98 To: ;tag=aprq1jcfl11-75bmme000g12f Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1777 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:08:55] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1' Method: REGISTER [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK45c60339;rport=5060 From: ;tag=as417b63cf To: ;tag=aprq1jcfl11-0cc8e7100g12f Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1777 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:08:55] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:08:55] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1' Method: REGISTER [Mar 7 23:08:57] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> BYE sip:8001@192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 76.8.29.198:5060;branch=z9hG4bK13cava20b801l5r882e1sd0g10n13.1 From: ;tag=gK04831af0 To: ;tag=as49be75cb Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 422083 BYE Max-Forwards: 66 Content-Length: 0 X-Remote-IP: 208.69.83.28 <-------------> [Mar 7 23:08:57] VERBOSE[31926] chan_sip.c: --- (9 headers 0 lines) --- [Mar 7 23:08:57] VERBOSE[31926][C-0000032d] chan_sip.c: Sending to 76.8.29.198:5060 (NAT) [Mar 7 23:08:57] VERBOSE[31926][C-0000032d] chan_sip.c: Scheduling destruction of SIP dialog '2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060' in 32000 ms (Method: BYE) [Mar 7 23:08:57] VERBOSE[31926][C-0000032d] chan_sip.c: <--- Transmitting (NAT) to 76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 76.8.29.198:5060;branch=z9hG4bK13cava20b801l5r882e1sd0g10n13.1;received=76.8.29.198;rport=5060 From: ;tag=gK04831af0 To: ;tag=as49be75cb Call-ID: 2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060 CSeq: 422083 BYE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Mar 7 23:08:57] VERBOSE[20478][C-0000032d] chan_sip.c: Scheduling destruction of SIP dialog '179d9786@192.168.1.107' in 32000 ms (Method: INVITE) [Mar 7 23:08:58] VERBOSE[31926] chan_sip.c: Retransmitting #6 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:09:01] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:3.212.223.16:52328 ---> <-------------> [Mar 7 23:09:02] VERBOSE[31926] chan_sip.c: Retransmitting #7 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:09:02] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '226e6685626415586581880k6023rmwp' Method: REGISTER [Mar 7 23:09:06] VERBOSE[31926] chan_sip.c: Retransmitting #8 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:09:10] VERBOSE[31926] chan_sip.c: Retransmitting #9 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:09:14] VERBOSE[31926] chan_sip.c: Retransmitting #10 (NAT) to 67.86.108.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-2903c85c;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as33db6584 Call-ID: 179d9786@192.168.1.107 CSeq: 8002 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 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 785419679 785419679 IN IP4 192.168.1.1 s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 c=IN IP4 192.168.1.1 t=0 0 m=audio 11812 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Mar 7 23:09:14] WARNING[31926] chan_sip.c: Retransmission timeout reached on transmission 179d9786@192.168.1.107 for seqno 8002 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 7 23:09:14] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '179d9786@192.168.1.107' Method: INVITE [Mar 7 23:09:15] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_Aron@sip.bulkvs.com [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK23b1c793;rport Max-Forwards: 70 From: ;tag=as76317b98 To: Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1778 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_Aron", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072b77ac15235ecbe96ac995e872c1d5e7d1e", response="67a0546c78e4b0f752c7171177ce195f" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:09:15] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_770@sip.bulkvs.com [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK733643ad;rport Max-Forwards: 70 From: ;tag=as417b63cf To: Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1778 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_770", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072ba682d84a98f217fadee0842ef8bf83322", response="c924ff52bf866d6e7b51a02d3a9fdd1a" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK23b1c793;rport=5060 From: ;tag=as76317b98 To: ;tag=aprq1jcfl11-75bmme000g14f Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1778 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:09:15] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1' Method: REGISTER [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK733643ad;rport=5060 From: ;tag=as417b63cf To: ;tag=aprq1jcfl11-0cc8e7100g14f Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1778 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:09:15] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:09:15] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1' Method: REGISTER [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> REGISTER sip:aron.com:5060 SIP/2.0 Call-ID: 2a91e9d1@192.168.1.107 Content-Length: 0 CSeq: 88088 REGISTER From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-7de57338;rport Authorization: DIGEST algorithm=MD5,nonce="2e4fac1f",realm="asterisk",response="630dfb8baaa3e4d84c80a20161ba2784",uri="sip:aron.com:5060",username="8001" User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:09:16] NOTICE[31926] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=SP1f9a94953c768237' [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-7de57338;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as26d49318 Call-ID: 2a91e9d1@192.168.1.107 CSeq: 88088 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2bf2c619", stale=true Content-Length: 0 <------------> [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '2a91e9d1@192.168.1.107' in 32000 ms (Method: REGISTER) [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:67.86.108.167:5060 ---> REGISTER sip:aron.com:5060 SIP/2.0 Call-ID: 2a91e9d1@192.168.1.107 Content-Length: 0 CSeq: 88089 REGISTER From: ;tag=SP1f9a94953c768237 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-32de8cd6;rport Authorization: DIGEST algorithm=MD5,nonce="2bf2c619",realm="asterisk",response="0b730e7a7a38eb1edc60a92c83726e3d",uri="sip:aron.com:5060",username="8001" User-Agent: OBIHAI/OBi110-1.3.0.2860 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: Sending to 67.86.108.167:5060 (NAT) [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 67.86.108.167:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-32de8cd6;received=67.86.108.167;rport=5060 From: ;tag=SP1f9a94953c768237 To: ;tag=as26d49318 Call-ID: 2a91e9d1@192.168.1.107 CSeq: 88089 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 08 Mar 2022 04:09:16 GMT Content-Length: 0 <------------> [Mar 7 23:09:16] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '2a91e9d1@192.168.1.107' in 32000 ms (Method: REGISTER) [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:192.168.1.44:5060 ---> REGISTER sip:192.168.1.1:5060 SIP/2.0 Call-ID: 8a3ae283@192.168.2.112 Content-Length: 0 CSeq: 42216 REGISTER From: ;tag=SP15a93efec3cd4da42 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-72bb6b2c;rport Authorization: DIGEST algorithm=MD5,nonce="135d94f3",realm="asterisk",response="3e031510ef652eda2ca71fa70434e1cd",uri="sip:192.168.1.1:5060",username="8180" User-Agent: OBIHAI/OBi110-1.3.0.2886 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: Sending to 192.168.1.44:5060 (NAT) [Mar 7 23:09:17] NOTICE[31926] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=SP15a93efec3cd4da42' [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.44:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-72bb6b2c;received=192.168.1.44;rport=5060 From: ;tag=SP15a93efec3cd4da42 To: ;tag=as34bf175f Call-ID: 8a3ae283@192.168.2.112 CSeq: 42216 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ff78c32", stale=true Content-Length: 0 <------------> [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '8a3ae283@192.168.2.112' in 32000 ms (Method: REGISTER) [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:192.168.1.44:5060 ---> REGISTER sip:192.168.1.1:5060 SIP/2.0 Call-ID: 8a3ae283@192.168.2.112 Content-Length: 0 CSeq: 42217 REGISTER From: ;tag=SP15a93efec3cd4da42 Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-24043649;rport Authorization: DIGEST algorithm=MD5,nonce="6ff78c32",realm="asterisk",response="a712b5abb705d438f366489da65102b5",uri="sip:192.168.1.1:5060",username="8180" User-Agent: OBIHAI/OBi110-1.3.0.2886 Contact: ;expires=60;+sip.instance="" Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Supported: replaces <-------------> [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: --- (13 headers 0 lines) --- [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: Sending to 192.168.1.44:5060 (NAT) [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.112:5060;branch=z9hG4bK-24043649;received=192.168.1.44;rport=5060 From: ;tag=SP15a93efec3cd4da42 To: ;tag=as34bf175f Call-ID: 8a3ae283@192.168.2.112 CSeq: 42217 REGISTER Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 08 Mar 2022 04:09:17 GMT Content-Length: 0 <------------> [Mar 7 23:09:17] VERBOSE[31926] chan_sip.c: Scheduling destruction of SIP dialog '8a3ae283@192.168.2.112' in 32000 ms (Method: REGISTER) [Mar 7 23:09:29] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '2de61aca2f74e13f1775fdc83146df5a@192.168.1.1:5060' Method: BYE [Mar 7 23:09:31] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:3.212.223.16:52328 ---> <-------------> [Mar 7 23:09:34] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:141.98.11.8:63691 ---> INVITE sip:002146520458262@123.333.1.1 SIP/2.0 Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887 Max-Forwards: 70 From: ;tag=851089322 To: Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: 207 Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH v=0 o=607 16264 18299 IN IP4 192.168.1.83 s=call c=IN IP4 192.168.1.83 t=0 0 m=audio 25282 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Mar 7 23:09:34] VERBOSE[31926] chan_sip.c: --- (11 headers 10 lines) --- [Mar 7 23:09:34] VERBOSE[31926] chan_sip.c: Sending to 141.98.11.8:63691 (NAT) [Mar 7 23:09:34] VERBOSE[31926][C-0000032e] chan_sip.c: Sending to 141.98.11.8:63691 (NAT) [Mar 7 23:09:34] VERBOSE[31926][C-0000032e] chan_sip.c: Using INVITE request as basis request - 402767820-1914731299-1494891811 [Mar 7 23:09:34] VERBOSE[31926][C-0000032e] chan_sip.c: No matching peer for '607' from '141.98.11.8:63691' [Mar 7 23:09:34] VERBOSE[31926][C-0000032e] chan_sip.c: <--- Reliably Transmitting (NAT) to 141.98.11.8:63691 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887;received=141.98.11.8;rport=63691 From: ;tag=851089322 To: ;tag=as698a68d3 Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35f27e78" Content-Length: 0 <------------> [Mar 7 23:09:34] VERBOSE[31926][C-0000032e] chan_sip.c: Scheduling destruction of SIP dialog '402767820-1914731299-1494891811' in 32000 ms (Method: INVITE) [Mar 7 23:09:34] VERBOSE[31926] chan_sip.c: Retransmitting #1 (NAT) to 141.98.11.8:63691: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887;received=141.98.11.8;rport=63691 From: ;tag=851089322 To: ;tag=as698a68d3 Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35f27e78" Content-Length: 0 --- [Mar 7 23:09:35] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_Aron@sip.bulkvs.com [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK49b12005;rport Max-Forwards: 70 From: ;tag=as76317b98 To: Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1779 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_Aron", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072b77ac15235ecbe96ac995e872c1d5e7d1e", response="67a0546c78e4b0f752c7171177ce195f" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: Retransmitting #2 (NAT) to 141.98.11.8:63691: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887;received=141.98.11.8;rport=63691 From: ;tag=851089322 To: ;tag=as698a68d3 Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35f27e78" Content-Length: 0 --- [Mar 7 23:09:35] NOTICE[31926] chan_sip.c: -- Re-registration for 609172_770@sip.bulkvs.com [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: REGISTER 12 headers, 0 lines [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: Reliably Transmitting (NAT) to 76.8.29.198:5060: REGISTER sip:sip.bulkvs.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4ec18ae7;rport Max-Forwards: 70 From: ;tag=as417b63cf To: Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1779 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Authorization: Digest username="609172_770", realm="sip.bulkvs.com", algorithm=MD5, uri="sip:sip.bulkvs.com", nonce="6226d6e1000072ba682d84a98f217fadee0842ef8bf83322", response="c924ff52bf866d6e7b51a02d3a9fdd1a" Expires: 120 Contact: Content-Length: 0 --- [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK49b12005;rport=5060 From: ;tag=as76317b98 To: ;tag=aprq1jcfl11-75bmme000g16f Call-ID: 4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1 CSeq: 1779 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:09:35] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '4935a7ca7fba2714026c27c4187b5d3a@123.333.1.1' Method: REGISTER [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: <--- SIP read from UDP:76.8.29.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;received=123.333.1.1;branch=z9hG4bK4ec18ae7;rport=5060 From: ;tag=as417b63cf To: ;tag=aprq1jcfl11-0cc8e7100g16f Call-ID: 3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1 CSeq: 1779 REGISTER Contact: ;expires=30 <-------------> [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: --- (7 headers 0 lines) --- [Mar 7 23:09:35] NOTICE[31926] chan_sip.c: Outbound Registration: Expiry for sip.bulkvs.com is 25 sec (Scheduling reregistration in 20 s) [Mar 7 23:09:35] VERBOSE[31926] chan_sip.c: Really destroying SIP dialog '3a49c33974cb17ba2c9f9083093c0aea@123.333.1.1' Method: REGISTER [Mar 7 23:09:37] VERBOSE[31926] chan_sip.c: Retransmitting #3 (NAT) to 141.98.11.8:63691: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887;received=141.98.11.8;rport=63691 From: ;tag=851089322 To: ;tag=as698a68d3 Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35f27e78" Content-Length: 0 --- [Mar 7 23:09:41] VERBOSE[31926] chan_sip.c: Retransmitting #4 (NAT) to 141.98.11.8:63691: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 141.98.11.8:63691;branch=z9hG4bK925741887;received=141.98.11.8;rport=63691 From: ;tag=851089322 To: ;tag=as698a68d3 Call-ID: 402767820-1914731299-1494891811 CSeq: 1 INVITE Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35f27e78" Content-Length: 0 ---