[Jun 22 06:54:21] Asterisk 16.6.2 built by root @ cingulariti4 on a x86_64 running Linux on 2019-12-17 09:56:32 UTC [Jun 22 06:54:21] VERBOSE[13449] logger.c: Asterisk Queue Logger restarted [Jun 22 06:54:39] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5070: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK5da85aa9 Max-Forwards: 70 From: "asterisk" ;tag=as23fc99e8 To: Contact: Call-ID: 5ccf829d4bc41a1c42ae368c37d8022c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:39] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK5da85aa9;received=172.16.14.104;rport=5060 From: "asterisk" ;tag=as23fc99e8 To: ;tag=as1860001f Call-ID: 5ccf829d4bc41a1c42ae368c37d8022c@172.16.14.104:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Jun 22 06:54:39] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:39] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5ccf829d4bc41a1c42ae368c37d8022c@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:54:41] VERBOSE[15826] pbx_spool.c: Attempting call on SIP/3115/9619928627 for application queue(support-installer1) (Retry 1) [Jun 22 06:54:41] VERBOSE[15826] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 06:54:41] VERBOSE[15826] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 06:54:41] VERBOSE[15826] chan_sip.c: Audio is at 10696 [Jun 22 06:54:41] VERBOSE[15826] chan_sip.c: Adding codec ulaw to SDP [Jun 22 06:54:41] VERBOSE[15826] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 22 06:54:41] VERBOSE[15826] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[15826] dial.c: Called 3115/9619928627 [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #5 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 15.206.16.158:5060: OPTIONS sip:15.206.16.158 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK5537758e;rport Max-Forwards: 70 From: "asterisk" ;tag=as2541a7b2 To: Contact: Call-ID: 1f12db7a22bc7e8e0426b16c08efaf39@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:15.206.16.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;rport=5060;received=103.142.198.210;branch=z9hG4bK5537758e Call-ID: 1f12db7a22bc7e8e0426b16c08efaf39@0.0.0.0:5060 From: "asterisk" ;tag=as2541a7b2 To: ;tag=z9hG4bK5537758e CSeq: 102 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1592808881/69b17c6ec00276c782a97a2b32d6b0aa",opaque="3bc46a15665356c2",algorithm=md5,qop="auth" Server: Asterisk PBX 16.10.0 Content-Length: 0 <-------------> [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: --- (9 headers 0 lines) --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '1f12db7a22bc7e8e0426b16c08efaf39@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Retransmitting #6 (no NAT) to 172.16.15.151:5070: INVITE sip:9619928627@172.16.15.151:5070 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c Max-Forwards: 70 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Contact: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 283435445 283435445 IN IP4 172.16.14.104 s=Asterisk PBX 16.6.2 c=IN IP4 172.16.14.104 t=0 0 m=audio 10696 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 --- [Jun 22 06:54:41] WARNING[13409] chan_sip.c: Retransmission timeout reached on transmission 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 448ms with no response [Jun 22 06:54:41] WARNING[13409] chan_sip.c: Hanging up call 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Jun 22 06:54:41] NOTICE[15826] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jun 22 06:54:41] NOTICE[15826] pbx_spool.c: Queued call to SIP/3115/9619928627 expired without completion after 0 attempts [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 52.1.96.171:5070: OPTIONS sip:ast101.ownmail.com SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK51a27b9d;rport Max-Forwards: 70 From: "asterisk" ;tag=as49f67022 To: Contact: Call-ID: 5164828c50b7aeb9637fb6042c6632e5@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '2664e9e70f3438e24542684d7cc2945c@172.16.15.151' Method: INVITE [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK21a9dc7b Max-Forwards: 70 From: "asterisk" ;tag=as6f1a2f25 To: Contact: Call-ID: 551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 40.117.253.69:5060: OPTIONS sip:40.117.253.69 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK179a45ce;rport Max-Forwards: 70 From: "asterisk" ;tag=as3db800df To: Contact: Call-ID: 6fa47d735622335f478be7365c06e1a1@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK51a27b9d;received=103.142.198.210;rport=5060 From: "asterisk" ;tag=as49f67022 To: ;tag=as5d249bc4 Call-ID: 5164828c50b7aeb9637fb6042c6632e5@0.0.0.0:5060 CSeq: 102 OPTIONS Server: M101 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5164828c50b7aeb9637fb6042c6632e5@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:40.117.253.69:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;rport=5060;received=103.142.198.210;branch=z9hG4bK179a45ce Call-ID: 6fa47d735622335f478be7365c06e1a1@0.0.0.0:5060 From: "asterisk" ;tag=as3db800df To: ;tag=z9hG4bK179a45ce CSeq: 102 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1592808881/90ac96f00aeb31ecaa1625c318b77e06",opaque="7a225b8c61d87d69",algorithm=md5,qop="auth" Server: Asterisk PBX 15.7.1 Content-Length: 0 <-------------> [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: --- (9 headers 0 lines) --- [Jun 22 06:54:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '6fa47d735622335f478be7365c06e1a1@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:54:42] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK21a9dc7b Max-Forwards: 70 From: "asterisk" ;tag=as6f1a2f25 To: Contact: Call-ID: 551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:43] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> <-------------> [Jun 22 06:54:43] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK21a9dc7b Max-Forwards: 70 From: "asterisk" ;tag=as6f1a2f25 To: Contact: Call-ID: 551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:43] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6a76824c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c378804 To: Contact: Call-ID: 6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:44] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK21a9dc7b Max-Forwards: 70 From: "asterisk" ;tag=as6f1a2f25 To: Contact: Call-ID: 551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:44] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6a76824c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c378804 To: Contact: Call-ID: 6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:45] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK21a9dc7b Max-Forwards: 70 From: "asterisk" ;tag=as6f1a2f25 To: Contact: Call-ID: 551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:45] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '551ae3db374a39a1639aa9182624bccd@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:54:45] VERBOSE[13409] chan_sip.c: Retransmitting #2 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6a76824c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c378804 To: Contact: Call-ID: 6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:46] VERBOSE[13409] chan_sip.c: Retransmitting #3 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6a76824c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c378804 To: Contact: Call-ID: 6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:47] VERBOSE[13409] chan_sip.c: Retransmitting #4 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6a76824c;rport Max-Forwards: 70 From: "asterisk" ;tag=as3c378804 To: Contact: Call-ID: 6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:47] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '6be099a450060e3c7b859bb2315e3380@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jun 22 06:54:50] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> REGISTER sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:46797;rport;branch=z9hG4bKPjbf9b8c81-2854-41a3-9683-df20c33adc2a Route: Max-Forwards: 70 From: ;tag=c5ad3c32-c805-4d42-87e4-6a877351de54 To: Call-ID: cd5a8173-62c0-491e-9699-57362da35ef7 CSeq: 9154 REGISTER User-Agent: pjsip python Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Sending to 172.16.14.104:46797 (NAT) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Sending to 172.16.14.104:46797 (NAT) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.14.104:46797 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.14.104:46797;branch=z9hG4bKPjbf9b8c81-2854-41a3-9683-df20c33adc2a;received=172.16.14.104;rport=46797 From: ;tag=c5ad3c32-c805-4d42-87e4-6a877351de54 To: ;tag=as4438a6a4 Call-ID: cd5a8173-62c0-491e-9699-57362da35ef7 CSeq: 9154 REGISTER Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edb5470" Content-Length: 0 <------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog 'cd5a8173-62c0-491e-9699-57362da35ef7' in 32000 ms (Method: REGISTER) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> REGISTER sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:46797;rport;branch=z9hG4bKPje691b544-886e-4f37-9701-6b7d7edf2535 Route: Max-Forwards: 70 From: ;tag=c5ad3c32-c805-4d42-87e4-6a877351de54 To: Call-ID: cd5a8173-62c0-491e-9699-57362da35ef7 CSeq: 9155 REGISTER User-Agent: pjsip python Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="9006", realm="asterisk", nonce="1edb5470", uri="sip:172.16.14.104", response="c4fe3a4578133461afb31ca66e54d225", algorithm=MD5 Content-Length: 0 <-------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: --- (14 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Sending to 172.16.14.104:46797 (NAT) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 172.16.14.104:46797: OPTIONS sip:9006@172.16.14.104:46797;ob SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK37373c7e;rport Max-Forwards: 70 From: "asterisk" ;tag=as6d4d8d28 To: Contact: Call-ID: 5ff0eb3043d3c5997fda461e2526efb6@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.14.104:46797 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:46797;branch=z9hG4bKPje691b544-886e-4f37-9701-6b7d7edf2535;received=172.16.14.104;rport=46797 From: ;tag=c5ad3c32-c805-4d42-87e4-6a877351de54 To: ;tag=as4438a6a4 Call-ID: cd5a8173-62c0-491e-9699-57362da35ef7 CSeq: 9155 REGISTER Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Mon, 22 Jun 2020 06:54:53 GMT Content-Length: 0 <------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060' in 320 ms (Method: NOTIFY) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 172.16.14.104:46797: NOTIFY sip:9006@172.16.14.104:46797;ob SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK1b381452;rport Max-Forwards: 70 From: "asterisk" ;tag=as794f281f To: Contact: Call-ID: 7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.6.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@172.16.14.104 Voice-Message: 0/0 (0/0) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog 'cd5a8173-62c0-491e-9699-57362da35ef7' in 32000 ms (Method: REGISTER) [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;rport=5060;received=172.16.14.104;branch=z9hG4bK37373c7e Call-ID: 5ff0eb3043d3c5997fda461e2526efb6@172.16.14.104:5060 From: "asterisk" ;tag=as6d4d8d28 To: ;tag=z9hG4bK37373c7e CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: pjsip python Content-Length: 0 <-------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5ff0eb3043d3c5997fda461e2526efb6@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 172.16.14.104:46797: NOTIFY sip:9006@172.16.14.104:46797;ob SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK1b381452;rport Max-Forwards: 70 From: "asterisk" ;tag=as794f281f To: Contact: Call-ID: 7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 16.6.2 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@172.16.14.104 Voice-Message: 0/0 (0/0) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;rport=5060;received=172.16.14.104;branch=z9hG4bK1b381452 Call-ID: 7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060 From: "asterisk" ;tag=as794f281f To: ;tag=z9hG4bK1b381452 CSeq: 102 NOTIFY Content-Length: 0 <-------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: --- (7 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060' Method: NOTIFY [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;rport=5060;received=172.16.14.104;branch=z9hG4bK1b381452 Call-ID: 7f8647974316f4d7662c2fac69b39228@172.16.14.104:5060 From: "asterisk" ;tag=as794f281f To: ;tag=z9hG4bK1b381452 CSeq: 102 NOTIFY Content-Length: 0 <-------------> [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: --- (7 headers 0 lines) --- [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '741ff1547657373a6a914153216340d4@172.16.15.151:5070' Method: OPTIONS [Jun 22 06:54:53] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '2e11792d39e298034e96e2004ddce6e6@172.16.15.151:5070' Method: OPTIONS [Jun 22 06:54:54] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5100c7832b3ae1ea300dd99b7b10cc2c@172.16.15.151:5070' Method: OPTIONS [Jun 22 06:54:55] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK353a6ca5 Max-Forwards: 70 From: "asterisk" ;tag=as13946215 To: Contact: Call-ID: 5593ac3378284e52496c4918641e619c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:55] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '7ff406ac47cf1098130d03242d93b149@172.16.15.151:5070' Method: OPTIONS [Jun 22 06:54:55] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '4036a6fb21498eb73fe83db060a11e9c@172.16.15.151:5070' Method: OPTIONS [Jun 22 06:54:56] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK353a6ca5 Max-Forwards: 70 From: "asterisk" ;tag=as13946215 To: Contact: Call-ID: 5593ac3378284e52496c4918641e619c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:57] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK353a6ca5 Max-Forwards: 70 From: "asterisk" ;tag=as13946215 To: Contact: Call-ID: 5593ac3378284e52496c4918641e619c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:57] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK0117e9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as57f4d0d1 To: Contact: Call-ID: 2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:58] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK353a6ca5 Max-Forwards: 70 From: "asterisk" ;tag=as13946215 To: Contact: Call-ID: 5593ac3378284e52496c4918641e619c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:58] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK0117e9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as57f4d0d1 To: Contact: Call-ID: 2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:59] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK353a6ca5 Max-Forwards: 70 From: "asterisk" ;tag=as13946215 To: Contact: Call-ID: 5593ac3378284e52496c4918641e619c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:59] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5593ac3378284e52496c4918641e619c@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:54:59] VERBOSE[13409] chan_sip.c: Retransmitting #2 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK0117e9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as57f4d0d1 To: Contact: Call-ID: 2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:54:59] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:54:59] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:00] VERBOSE[13409] chan_sip.c: Retransmitting #3 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK0117e9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as57f4d0d1 To: Contact: Call-ID: 2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:00] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '085361875599dd9a7d44f7781df6554e@52.1.96.171:5070' Method: OPTIONS [Jun 22 06:55:01] VERBOSE[13409] chan_sip.c: Retransmitting #4 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK0117e9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as57f4d0d1 To: Contact: Call-ID: 2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:01] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '2053cf5866b3551c2eadd065396f1609@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:03] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:03] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:04] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:04] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:05] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:05] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:07] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:07] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:08] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> <-------------> [Jun 22 06:55:09] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7959ee95 Max-Forwards: 70 From: "asterisk" ;tag=as69e23871 To: Contact: Call-ID: 2c083e51703de96131633e25269a5d7f@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:10] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7959ee95 Max-Forwards: 70 From: "asterisk" ;tag=as69e23871 To: Contact: Call-ID: 2c083e51703de96131633e25269a5d7f@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:11] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:11] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:11] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7959ee95 Max-Forwards: 70 From: "asterisk" ;tag=as69e23871 To: Contact: Call-ID: 2c083e51703de96131633e25269a5d7f@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:11] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK21e469a4;rport Max-Forwards: 70 From: "asterisk" ;tag=as35a81b40 To: Contact: Call-ID: 7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:12] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7959ee95 Max-Forwards: 70 From: "asterisk" ;tag=as69e23871 To: Contact: Call-ID: 2c083e51703de96131633e25269a5d7f@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:12] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK21e469a4;rport Max-Forwards: 70 From: "asterisk" ;tag=as35a81b40 To: Contact: Call-ID: 7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7959ee95 Max-Forwards: 70 From: "asterisk" ;tag=as69e23871 To: Contact: Call-ID: 2c083e51703de96131633e25269a5d7f@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '2c083e51703de96131633e25269a5d7f@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Retransmitting #2 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK21e469a4;rport Max-Forwards: 70 From: "asterisk" ;tag=as35a81b40 To: Contact: Call-ID: 7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:13] NOTICE[13409] chan_sip.c: -- Re-registration for 1555036869@ast101.ownmail.com [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: REGISTER 12 headers, 0 lines [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 52.1.96.171:5070: REGISTER sip:ast101.ownmail.com:5070 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK4db8a35f;rport Max-Forwards: 70 From: ;tag=as1f58551b To: Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 120 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 16.6.2 Authorization: Digest username="1555036869", realm="52.1.96.171:5070", algorithm=MD5, uri="sip:ast101.ownmail.com:5070", nonce="33b82657", response="feb9fb4cd4924f45abae7c06b73d8001" Expires: 120 Contact: Content-Length: 0 --- [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK4db8a35f;received=103.142.198.210;rport=5060 From: ;tag=as1f58551b To: ;tag=as4278d18e Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 120 REGISTER Server: M101 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="52.1.96.171:5070", nonce="48dda773" Content-Length: 0 <-------------> [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: --- (11 headers 0 lines) --- [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Responding to challenge, registration to domain/host name ast101.ownmail.com [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: REGISTER 12 headers, 0 lines [Jun 22 06:55:13] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 52.1.96.171:5070: REGISTER sip:ast101.ownmail.com SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK52871fc4;rport Max-Forwards: 70 From: ;tag=as1f58551b To: Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 121 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 16.6.2 Authorization: Digest username="1555036869", realm="52.1.96.171:5070", algorithm=MD5, uri="sip:ast101.ownmail.com:5070", nonce="48dda773", response="c016367d6507b73d7371a23715a3a67e" Expires: 120 Contact: Content-Length: 0 --- [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK52871fc4;received=103.142.198.210;rport=5060 From: ;tag=as1f58551b To: ;tag=as3b512ec3 Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 121 REGISTER Server: M101 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="52.1.96.171:5070", nonce="3fc82027" Content-Length: 0 <-------------> [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: --- (11 headers 0 lines) --- [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Responding to challenge, registration to domain/host name ast101.ownmail.com [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: REGISTER 12 headers, 0 lines [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 52.1.96.171:5070: REGISTER sip:ast101.ownmail.com SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7f423ce8;rport Max-Forwards: 70 From: ;tag=as1f58551b To: Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 122 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 16.6.2 Authorization: Digest username="1555036869", realm="52.1.96.171:5070", algorithm=MD5, uri="sip:ast101.ownmail.com", nonce="3fc82027", response="4646967007194e14df2fce8b1d01db7e" Expires: 120 Contact: Content-Length: 0 --- [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> OPTIONS sip:s@0.0.0.0:5060 SIP/2.0 Via: SIP/2.0/UDP 52.1.96.171:5070;branch=z9hG4bK77cb6df7;rport Max-Forwards: 70 From: "asterisk" ;tag=as630f35ee To: Contact: Call-ID: 4d868184651e2f85126be233256e99f6@52.1.96.171:5070 CSeq: 102 OPTIONS User-Agent: M101 Date: Mon, 22 Jun 2020 06:55:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Sending to 52.1.96.171:5070 (NAT) [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 0.0.0.0) [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 52.1.96.171:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 52.1.96.171:5070;branch=z9hG4bK77cb6df7;received=52.1.96.171;rport=5070 From: "asterisk" ;tag=as630f35ee To: ;tag=as4ea4e280 Call-ID: 4d868184651e2f85126be233256e99f6@52.1.96.171:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '4d868184651e2f85126be233256e99f6@52.1.96.171:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7f423ce8;received=103.142.198.210;rport=5060 From: ;tag=as1f58551b To: ;tag=as3b512ec3 Call-ID: 4429e28b1e46d9655a8a70d2450a3672@127.0.1.1 CSeq: 122 REGISTER Server: M101 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 22 Jun 2020 06:55:14 GMT Content-Length: 0 <-------------> [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:14] NOTICE[13409] chan_sip.c: Outbound Registration: Expiry for ast101.ownmail.com is 120 sec (Scheduling reregistration in 105 s) [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '4429e28b1e46d9655a8a70d2450a3672@127.0.1.1' Method: REGISTER [Jun 22 06:55:14] VERBOSE[13409] chan_sip.c: Retransmitting #3 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK21e469a4;rport Max-Forwards: 70 From: "asterisk" ;tag=as35a81b40 To: Contact: Call-ID: 7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:15] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:15] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:15] VERBOSE[13409] chan_sip.c: Retransmitting #4 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK21e469a4;rport Max-Forwards: 70 From: "asterisk" ;tag=as35a81b40 To: Contact: Call-ID: 7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:15] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '7fabf40969d7db2d37d972d7335b69a9@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:19] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:19] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> OPTIONS sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK724cb408;rport Max-Forwards: 70 From: "asterisk" ;tag=as623d15ca To: Contact: Call-ID: 7488fb12435ded0e08f84efd22c5c607@172.16.15.151:5070 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.29.1 Date: Mon, 22 Jun 2020 07:01:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 172.16.14.104) [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK724cb408;received=172.16.15.151;rport=5070 From: "asterisk" ;tag=as623d15ca To: ;tag=as28ebfe40 Call-ID: 7488fb12435ded0e08f84efd22c5c607@172.16.15.151:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '7488fb12435ded0e08f84efd22c5c607@172.16.15.151:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> OPTIONS sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK2d862f5c;rport Max-Forwards: 70 From: "asterisk" ;tag=as17b4f7cd To: Contact: Call-ID: 7034bbc651113be659372d3b24bd1d9b@172.16.15.151:5070 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.29.1 Date: Mon, 22 Jun 2020 07:01:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 172.16.14.104) [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK2d862f5c;received=172.16.15.151;rport=5070 From: "asterisk" ;tag=as17b4f7cd To: ;tag=as1bae1281 Call-ID: 7034bbc651113be659372d3b24bd1d9b@172.16.15.151:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:21] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '7034bbc651113be659372d3b24bd1d9b@172.16.15.151:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> OPTIONS sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK0a8e310f;rport Max-Forwards: 70 From: "asterisk" ;tag=as22f34d06 To: Contact: Call-ID: 32ee4a3207d5d88517c0b117376d253a@172.16.15.151:5070 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.29.1 Date: Mon, 22 Jun 2020 07:01:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 172.16.14.104) [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK0a8e310f;received=172.16.15.151;rport=5070 From: "asterisk" ;tag=as22f34d06 To: ;tag=as35ad00ab Call-ID: 32ee4a3207d5d88517c0b117376d253a@172.16.15.151:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:22] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '32ee4a3207d5d88517c0b117376d253a@172.16.15.151:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> <-------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK314549c3 Max-Forwards: 70 From: "asterisk" ;tag=as3031fff7 To: Contact: Call-ID: 06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> OPTIONS sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK676f8a04;rport Max-Forwards: 70 From: "asterisk" ;tag=as16f8a38c To: Contact: Call-ID: 703d45bf0873b1c116b0818928753b79@172.16.15.151:5070 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.29.1 Date: Mon, 22 Jun 2020 07:01:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 172.16.14.104) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK676f8a04;received=172.16.15.151;rport=5070 From: "asterisk" ;tag=as16f8a38c To: ;tag=as12fca7ad Call-ID: 703d45bf0873b1c116b0818928753b79@172.16.15.151:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '703d45bf0873b1c116b0818928753b79@172.16.15.151:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> OPTIONS sip:172.16.14.104 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK774ab2e2;rport Max-Forwards: 70 From: "asterisk" ;tag=as714e5545 To: Contact: Call-ID: 035ed47f5bd22a547a107f5a605dbcc0@172.16.15.151:5070 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.29.1 Date: Mon, 22 Jun 2020 07:01:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: --- (13 headers 0 lines) --- [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Looking for s in default (domain 172.16.14.104) [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK774ab2e2;received=172.16.15.151;rport=5070 From: "asterisk" ;tag=as714e5545 To: ;tag=as06abfb0c Call-ID: 035ed47f5bd22a547a107f5a605dbcc0@172.16.15.151:5070 CSeq: 102 OPTIONS Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Jun 22 06:55:23] VERBOSE[13409] chan_sip.c: Scheduling destruction of SIP dialog '035ed47f5bd22a547a107f5a605dbcc0@172.16.15.151:5070' in 32000 ms (Method: OPTIONS) [Jun 22 06:55:24] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK314549c3 Max-Forwards: 70 From: "asterisk" ;tag=as3031fff7 To: Contact: Call-ID: 06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:25] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog 'cd5a8173-62c0-491e-9699-57362da35ef7' Method: REGISTER [Jun 22 06:55:25] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK314549c3 Max-Forwards: 70 From: "asterisk" ;tag=as3031fff7 To: Contact: Call-ID: 06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:25] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6e57ebdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e626772 To: Contact: Call-ID: 0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:26] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK314549c3 Max-Forwards: 70 From: "asterisk" ;tag=as3031fff7 To: Contact: Call-ID: 06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:26] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6e57ebdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e626772 To: Contact: Call-ID: 0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:27] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:27] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:27] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK314549c3 Max-Forwards: 70 From: "asterisk" ;tag=as3031fff7 To: Contact: Call-ID: 06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:27] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '06c249be385efa771b7b89a72bb1ce0c@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:55:27] VERBOSE[13409] chan_sip.c: Retransmitting #2 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6e57ebdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e626772 To: Contact: Call-ID: 0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:28] VERBOSE[13409] chan_sip.c: Retransmitting #3 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6e57ebdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e626772 To: Contact: Call-ID: 0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:29] VERBOSE[13409] chan_sip.c: Retransmitting #4 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK6e57ebdc;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e626772 To: Contact: Call-ID: 0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:29] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '0332f1dd21121f970586517b6e03ffe0@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:31] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:31] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK730afc3c;received=172.16.14.104;rport=5060 From: "9619928627 'I want to deposit money'" ;tag=as313b27fa To: ;tag=as63edeec9 Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 INVITE Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 240 v=0 o=root 255657849 255657849 IN IP4 172.16.15.151 s=Asterisk PBX 13.29.1 c=IN IP4 172.16.15.151 t=0 0 m=audio 17072 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 <-------------> [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: --- (14 headers 11 lines) --- [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> BYE sip:3115@172.16.14.104:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK1c76b737;rport Max-Forwards: 70 From: ;tag=as63edeec9 To: "9619928627 'I want to deposit money'" ;tag=as313b27fa Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 BYE User-Agent: Asterisk PBX 13.29.1 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 <-------------> [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: --- (11 headers 0 lines) --- [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: Sending to 172.16.15.151:5070 (NAT) [Jun 22 06:55:35] VERBOSE[13409] chan_sip.c: <--- Transmitting (NAT) to 172.16.15.151:5070 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 172.16.15.151:5070;branch=z9hG4bK1c76b737;received=172.16.15.151;rport=5070 From: ;tag=as63edeec9 To: "9619928627 'I want to deposit money'" ;tag=as313b27fa Call-ID: 2664e9e70f3438e24542684d7cc2945c@172.16.15.151 CSeq: 102 BYE Server: Asterisk PBX 16.6.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jun 22 06:55:37] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK4a85e0e8 Max-Forwards: 70 From: "asterisk" ;tag=as1dd2ccdd To: Contact: Call-ID: 21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:38] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.14.104:46797 ---> <-------------> [Jun 22 06:55:38] VERBOSE[13409] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK4a85e0e8 Max-Forwards: 70 From: "asterisk" ;tag=as1dd2ccdd To: Contact: Call-ID: 21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: Retransmitting #2 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK4a85e0e8 Max-Forwards: 70 From: "asterisk" ;tag=as1dd2ccdd To: Contact: Call-ID: 21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7207fc54;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ac737cb To: Contact: Call-ID: 520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.15.151:5070: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7b439666 Max-Forwards: 70 From: "asterisk" ;tag=as4cd47e99 To: Contact: Call-ID: 5a830d7136433e9c731626d648685de1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:172.16.15.151:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK7b439666;received=172.16.14.104;rport=5060 From: "asterisk" ;tag=as4cd47e99 To: ;tag=as5a053996 Call-ID: 5a830d7136433e9c731626d648685de1@172.16.14.104:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.29.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:55:39] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '5a830d7136433e9c731626d648685de1@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:55:40] VERBOSE[13409] chan_sip.c: Retransmitting #3 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK4a85e0e8 Max-Forwards: 70 From: "asterisk" ;tag=as1dd2ccdd To: Contact: Call-ID: 21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:40] VERBOSE[13409] chan_sip.c: Retransmitting #1 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7207fc54;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ac737cb To: Contact: Call-ID: 520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 15.206.16.158:5060: OPTIONS sip:15.206.16.158 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK09aee791;rport Max-Forwards: 70 From: "asterisk" ;tag=as60e5f395 To: Contact: Call-ID: 0bcd52766c7804756c967397391b9add@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:15.206.16.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;rport=5060;received=103.142.198.210;branch=z9hG4bK09aee791 Call-ID: 0bcd52766c7804756c967397391b9add@0.0.0.0:5060 From: "asterisk" ;tag=as60e5f395 To: ;tag=z9hG4bK09aee791 CSeq: 102 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1592808941/7ae947968edfbca522e7e2aeb246a5ca",opaque="0317d0881c8e1454",algorithm=md5,qop="auth" Server: Asterisk PBX 16.10.0 Content-Length: 0 <-------------> [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: --- (9 headers 0 lines) --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '0bcd52766c7804756c967397391b9add@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Retransmitting #4 (no NAT) to 172.16.15.151:5060: OPTIONS sip:172.16.15.151 SIP/2.0 Via: SIP/2.0/UDP 172.16.14.104:5060;branch=z9hG4bK4a85e0e8 Max-Forwards: 70 From: "asterisk" ;tag=as1dd2ccdd To: Contact: Call-ID: 21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '21faf2753b3bdf713706ba722bccaef1@172.16.14.104:5060' Method: OPTIONS [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Retransmitting #2 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7207fc54;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ac737cb To: Contact: Call-ID: 520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 52.1.96.171:5070: OPTIONS sip:ast101.ownmail.com SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK3b161de2;rport Max-Forwards: 70 From: "asterisk" ;tag=as0dcc79c5 To: Contact: Call-ID: 36d4f7dd5660376403dd34fa797bf811@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:41] VERBOSE[13409] chan_sip.c: Reliably Transmitting (NAT) to 40.117.253.69:5060: OPTIONS sip:40.117.253.69 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK36d4cbf0;rport Max-Forwards: 70 From: "asterisk" ;tag=as75bf5c2c To: Contact: Call-ID: 0ad68f4e2131a863708943b00471deb0@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:52.1.96.171:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK3b161de2;received=103.142.198.210;rport=5060 From: "asterisk" ;tag=as0dcc79c5 To: ;tag=as64d3e224 Call-ID: 36d4f7dd5660376403dd34fa797bf811@0.0.0.0:5060 CSeq: 102 OPTIONS Server: M101 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: --- (12 headers 0 lines) --- [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '36d4f7dd5660376403dd34fa797bf811@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: <--- SIP read from UDP:40.117.253.69:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;rport=5060;received=103.142.198.210;branch=z9hG4bK36d4cbf0 Call-ID: 0ad68f4e2131a863708943b00471deb0@0.0.0.0:5060 From: "asterisk" ;tag=as75bf5c2c To: ;tag=z9hG4bK36d4cbf0 CSeq: 102 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1592808942/512157aba1fe7f9ea58d4f265ff19cc4",opaque="10c6aecb2ef86a7c",algorithm=md5,qop="auth" Server: Asterisk PBX 15.7.1 Content-Length: 0 <-------------> [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: --- (9 headers 0 lines) --- [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '0ad68f4e2131a863708943b00471deb0@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:55:42] VERBOSE[13409] chan_sip.c: Retransmitting #3 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7207fc54;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ac737cb To: Contact: Call-ID: 520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:43] VERBOSE[13409] chan_sip.c: Retransmitting #4 (NAT) to 3.6.232.225:5060: OPTIONS sip:3.6.232.225 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7207fc54;rport Max-Forwards: 70 From: "asterisk" ;tag=as3ac737cb To: Contact: Call-ID: 520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 16.6.2 Date: Mon, 22 Jun 2020 06:55:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 22 06:55:43] VERBOSE[13409] chan_sip.c: Really destroying SIP dialog '520f168958fe572133ac7ba90063c6ff@0.0.0.0:5060' Method: OPTIONS [Jun 22 06:56:46] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:32881 [Jun 22 06:56:46] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms) [Jun 22 06:57:04] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:36954 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP learning after remote address set to: 172.16.15.151:13596 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000014", "02268388081") in new stack [Jun 22 06:57:50] VERBOSE[15941][C-00000008] app_verbose.c: 02268388081 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000014", "number=8081") in new stack [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000014", "8081") in new stack [Jun 22 06:57:50] VERBOSE[15941][C-00000008] app_verbose.c: 8081 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000014", "1?kafka:") in new stack [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000014", "** Channel SIP/02268388081-00000014 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 06:57:50] WARNING[15941][C-00000008] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592809070EPOCH'? [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000014", "22062020-06:57:50-02268388081-customer-incoming.wav") in new stack [Jun 22 06:57:50] VERBOSE[15941][C-00000008] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000014", "support-installer3") in new stack [Jun 22 06:57:50] VERBOSE[15942][C-00000008] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000014 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000014' [Jun 22 06:57:50] VERBOSE[15941][C-00000008] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] app_queue.c: Called SIP/9010 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 06:57:50] VERBOSE[13409][C-00000008] res_rtp_asterisk.c: 0x7fd2bc01a1c0 -- Strict RTP learning after remote address set to: 172.16.14.104:33217 [Jun 22 06:57:50] VERBOSE[15941][C-00000008] app_queue.c: SIP/9010-00000015 answered SIP/02268388081-00000014 [Jun 22 06:57:50] NOTICE[15941][C-00000008] app_queue.c: Delaying member connect for 2 seconds [Jun 22 06:57:52] VERBOSE[15941][C-00000008] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 06:57:52] VERBOSE[15941][C-00000008] res_rtp_asterisk.c: 0x7fd2bc01a1c0 -- Strict RTP switching to RTP target address 172.16.14.104:33217 as source [Jun 22 06:57:53] VERBOSE[15941][C-00000008] file.c: Playing 'digits/3.ulaw' (language 'en') [Jun 22 06:57:54] VERBOSE[15941][C-00000008] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 06:57:55] VERBOSE[15941][C-00000008] res_rtp_asterisk.c: 0x7fd2bc01a1c0 -- Strict RTP learning complete - Locking on source address 172.16.14.104:33217 [Jun 22 06:57:55] VERBOSE[15941][C-00000008] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000014 [Jun 22 06:57:55] VERBOSE[15949][C-00000008] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000014 [Jun 22 06:57:55] VERBOSE[15952][C-00000008] bridge_channel.c: Channel SIP/9010-00000015 joined 'simple_bridge' basic-bridge <50c11e6c-5e85-4ac7-a3cc-a605710b3132> [Jun 22 06:57:55] VERBOSE[15941][C-00000008] bridge_channel.c: Channel SIP/02268388081-00000014 joined 'simple_bridge' basic-bridge <50c11e6c-5e85-4ac7-a3cc-a605710b3132> [Jun 22 06:57:56] VERBOSE[15941][C-00000008] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP switching to RTP target address 172.16.15.151:13596 as source [Jun 22 06:57:56] VERBOSE[15941][C-00000008] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP learning complete - Locking on source address 172.16.15.151:13596 [Jun 22 06:58:08] NOTICE[13409] chan_sip.c: Peer '9010' is now UNREACHABLE! Last qualify: 1 [Jun 22 06:58:40] VERBOSE[15941][C-00000008] bridge_channel.c: Channel SIP/02268388081-00000014 left 'simple_bridge' basic-bridge <50c11e6c-5e85-4ac7-a3cc-a605710b3132> [Jun 22 06:58:40] VERBOSE[15952][C-00000008] bridge_channel.c: Channel SIP/9010-00000015 left 'simple_bridge' basic-bridge <50c11e6c-5e85-4ac7-a3cc-a605710b3132> [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000014' [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000014", "GLOBAL(call_end_time)=2020-06-22 06:58:40") in new stack [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 06:58:40' [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000014", ""call Hangup details - - 4 - 30 - /var/spool/asterisk/monitor/1592809070.27.wav - 2020-06-22 06:57:50 - 2020-06-22 06:25:09 - 2020-06-22 06:58:40 - 2020-06-22 06:58:40 - "") in new stack [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000014", "CRSLT={ "status": 200 }") in new stack [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000014", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000014", "") in new stack [Jun 22 06:58:40] VERBOSE[15941][C-00000008] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000014' [Jun 22 06:58:40] VERBOSE[15942][C-00000008] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 06:58:40] VERBOSE[15942][C-00000008] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000014 [Jun 22 06:58:40] VERBOSE[15949][C-00000008] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 06:58:40] VERBOSE[15949][C-00000008] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000014 [Jun 22 07:00:37] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:37759 [Jun 22 07:00:37] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms) [Jun 22 07:01:26] VERBOSE[13409][C-00000009] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:01:26] VERBOSE[13409][C-00000009] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:01:26] VERBOSE[13409][C-00000009] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP learning after remote address set to: 172.16.15.151:19984 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000016", "02268388081") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] app_verbose.c: 02268388081 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000016", "number=8081") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000016", "8081") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] app_verbose.c: 8081 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000016", "1?kafka:") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000016", "** Channel SIP/02268388081-00000016 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:01:26] WARNING[16271][C-00000009] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592809286EPOCH'? [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000016", "22062020-07:01:26-02268388081-customer-incoming.wav") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000016", "support-installer3") in new stack [Jun 22 07:01:26] VERBOSE[16272][C-00000009] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000016 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000016' [Jun 22 07:01:26] VERBOSE[16271][C-00000009] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] app_queue.c: Called SIP/9010 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000016 [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000016' [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000016", "GLOBAL(call_end_time)=2020-06-22 07:01:26") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:01:26' [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000016", ""call Hangup details - - - 30 - /var/spool/asterisk/monitor/22062020-07:01:26-02268388081-customer-incoming.wav - 2020-06-22 07:01:26 - 2020-06-22 06:25:09 - 2020-06-22 07:01:26 - 2020-06-22 07:01:26 - "") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000016", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000016", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000016", "") in new stack [Jun 22 07:01:26] VERBOSE[16271][C-00000009] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000016' [Jun 22 07:01:26] VERBOSE[16272][C-00000009] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:01:26] VERBOSE[16272][C-00000009] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000016 [Jun 22 07:01:26] VERBOSE[13409][C-00000009] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:01:26] VERBOSE[13409][C-00000009] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:01:26] VERBOSE[13409][C-00000009] res_rtp_asterisk.c: 0x7fd360016680 -- Strict RTP learning after remote address set to: 172.16.14.104:52470 [Jun 22 07:01:41] NOTICE[13409] chan_sip.c: Peer '9010' is now UNREACHABLE! Last qualify: 1 [Jun 22 07:02:42] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:45050 [Jun 22 07:02:42] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms) [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP learning after remote address set to: 172.16.15.151:14800 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000018", "02268388081") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] app_verbose.c: 02268388081 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000018", "number=8081") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000018", "8081") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] app_verbose.c: 8081 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000018", "1?kafka:") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000018", "** Channel SIP/02268388081-00000018 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:05:13] WARNING[17018][C-0000000a] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592809513EPOCH'? [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000018", "22062020-07:05:13-02268388081-customer-incoming.wav") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000018", "support-installer3") in new stack [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000018' [Jun 22 07:05:13] VERBOSE[17019][C-0000000a] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000018 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] app_queue.c: Called SIP/9010 [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:05:13] VERBOSE[13409][C-0000000a] res_rtp_asterisk.c: 0x7fd2d0008e30 -- Strict RTP learning after remote address set to: 172.16.14.104:46471 [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] app_queue.c: SIP/9010-00000019 answered SIP/02268388081-00000018 [Jun 22 07:05:13] NOTICE[17018][C-0000000a] app_queue.c: Delaying member connect for 2 seconds [Jun 22 07:05:13] VERBOSE[17018][C-0000000a] res_rtp_asterisk.c: 0x7fd2d0008e30 -- Strict RTP switching to RTP target address 172.16.14.104:46471 as source [Jun 22 07:05:15] VERBOSE[17018][C-0000000a] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 07:05:16] VERBOSE[17018][C-0000000a] file.c: Playing 'digits/3.ulaw' (language 'en') [Jun 22 07:05:17] VERBOSE[17018][C-0000000a] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000018 [Jun 22 07:05:18] VERBOSE[17024][C-0000000a] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000018 [Jun 22 07:05:18] VERBOSE[17027][C-0000000a] bridge_channel.c: Channel SIP/9010-00000019 joined 'simple_bridge' basic-bridge <02c88ca8-b680-48b7-8070-dc8f0600525f> [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] bridge_channel.c: Channel SIP/02268388081-00000018 joined 'simple_bridge' basic-bridge <02c88ca8-b680-48b7-8070-dc8f0600525f> [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP switching to RTP target address 172.16.15.151:14800 as source [Jun 22 07:05:18] WARNING[17018][C-0000000a] channel.c: Unable to find a codec translation path: (g729) -> (ulaw) [Jun 22 07:05:18] WARNING[17018][C-0000000a] channel.c: Unable to find a codec translation path: (ulaw) -> (g729) [Jun 22 07:05:18] WARNING[17018][C-0000000a] translate.c: No translator path: (starting codec is not valid) [Jun 22 07:05:18] WARNING[17027][C-0000000a] channel.c: Unable to find a codec translation path: (g729) -> (ulaw) [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] res_rtp_asterisk.c: 0x7fd27805b9c0 -- Strict RTP learning complete - Locking on source address 172.16.15.151:14800 [Jun 22 07:05:18] WARNING[17018][C-0000000a] translate.c: No translator path: (starting codec is not valid) [Jun 22 07:05:18] WARNING[17018][C-0000000a] translate.c: No translator path: (starting codec is not valid) [Jun 22 07:05:18] WARNING[17018][C-0000000a] translate.c: No translator path: (starting codec is not valid) [Jun 22 07:05:18] VERBOSE[17027][C-0000000a] bridge_channel.c: Channel SIP/9010-00000019 left 'simple_bridge' basic-bridge <02c88ca8-b680-48b7-8070-dc8f0600525f> [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] bridge_channel.c: Channel SIP/02268388081-00000018 left 'simple_bridge' basic-bridge <02c88ca8-b680-48b7-8070-dc8f0600525f> [Jun 22 07:05:18] WARNING[17018][C-0000000a] channel.c: Unable to find a codec translation path: (g729) -> (ulaw) [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000018' [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000018", "GLOBAL(call_end_time)=2020-06-22 07:05:18") in new stack [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:05:18' [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000018", ""call Hangup details - - 5 - 30 - /var/spool/asterisk/monitor/1592809513.33.wav - 2020-06-22 07:05:13 - 2020-06-22 06:25:09 - 2020-06-22 07:05:18 - 2020-06-22 07:05:18 - "") in new stack [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000018", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000018", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000018", "") in new stack [Jun 22 07:05:18] VERBOSE[17018][C-0000000a] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000018' [Jun 22 07:05:18] VERBOSE[17019][C-0000000a] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:05:18] VERBOSE[17024][C-0000000a] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:05:18] VERBOSE[17024][C-0000000a] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000018 [Jun 22 07:05:18] VERBOSE[17019][C-0000000a] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000018 [Jun 22 07:05:28] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:40135 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] res_rtp_asterisk.c: 0x7fd27805d230 -- Strict RTP learning after remote address set to: 172.16.15.151:16210 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-0000001a", "02268388081") in new stack [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] app_verbose.c: 02268388081 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-0000001a", "number=8081") in new stack [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-0000001a", "8081") in new stack [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] app_verbose.c: 8081 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-0000001a", "1?kafka:") in new stack [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-0000001a", "** Channel SIP/02268388081-0000001a queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:05:49] WARNING[17082][C-0000000b] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592809549EPOCH'? [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-0000001a", "22062020-07:05:49-02268388081-customer-incoming.wav") in new stack [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-0000001a", "support-installer3") in new stack [Jun 22 07:05:49] VERBOSE[17083][C-0000000b] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-0000001a [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-0000001a' [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] app_queue.c: Called SIP/9010 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:05:49] VERBOSE[13409][C-0000000b] res_rtp_asterisk.c: 0x7fd2bc02a950 -- Strict RTP learning after remote address set to: 172.16.14.104:42108 [Jun 22 07:05:49] VERBOSE[17082][C-0000000b] app_queue.c: SIP/9010-0000001b answered SIP/02268388081-0000001a [Jun 22 07:05:49] NOTICE[17082][C-0000000b] app_queue.c: Delaying member connect for 2 seconds [Jun 22 07:05:51] VERBOSE[17082][C-0000000b] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 07:05:51] VERBOSE[17082][C-0000000b] res_rtp_asterisk.c: 0x7fd2bc02a950 -- Strict RTP switching to RTP target address 172.16.14.104:42108 as source [Jun 22 07:05:52] VERBOSE[17082][C-0000000b] file.c: Playing 'digits/3.ulaw' (language 'en') [Jun 22 07:05:53] VERBOSE[17082][C-0000000b] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 07:05:54] VERBOSE[17082][C-0000000b] res_rtp_asterisk.c: 0x7fd2bc02a950 -- Strict RTP learning complete - Locking on source address 172.16.14.104:42108 [Jun 22 07:05:54] VERBOSE[17082][C-0000000b] res_musiconhold.c: Stopped music on hold on SIP/02268388081-0000001a [Jun 22 07:05:54] VERBOSE[17090][C-0000000b] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-0000001a [Jun 22 07:05:54] VERBOSE[17093][C-0000000b] bridge_channel.c: Channel SIP/9010-0000001b joined 'simple_bridge' basic-bridge <5b9cabb6-377d-4f39-89c2-2ce0c6280ddf> [Jun 22 07:05:54] VERBOSE[17082][C-0000000b] bridge_channel.c: Channel SIP/02268388081-0000001a joined 'simple_bridge' basic-bridge <5b9cabb6-377d-4f39-89c2-2ce0c6280ddf> [Jun 22 07:05:54] VERBOSE[17082][C-0000000b] res_rtp_asterisk.c: 0x7fd27805d230 -- Strict RTP switching to RTP target address 172.16.15.151:16210 as source [Jun 22 07:05:54] VERBOSE[17082][C-0000000b] res_rtp_asterisk.c: 0x7fd27805d230 -- Strict RTP learning complete - Locking on source address 172.16.15.151:16210 [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] bridge_channel.c: Channel SIP/02268388081-0000001a left 'simple_bridge' basic-bridge <5b9cabb6-377d-4f39-89c2-2ce0c6280ddf> [Jun 22 07:06:15] VERBOSE[17093][C-0000000b] bridge_channel.c: Channel SIP/9010-0000001b left 'simple_bridge' basic-bridge <5b9cabb6-377d-4f39-89c2-2ce0c6280ddf> [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-0000001a' [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-0000001a", "GLOBAL(call_end_time)=2020-06-22 07:06:15") in new stack [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:06:15' [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-0000001a", ""call Hangup details - - 6 - 21 - /var/spool/asterisk/monitor/1592809549.36.wav - 2020-06-22 07:05:49 - 2020-06-22 06:25:09 - 2020-06-22 07:06:15 - 2020-06-22 07:06:15 - "") in new stack [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-0000001a", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-0000001a", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-0000001a", "") in new stack [Jun 22 07:06:15] VERBOSE[17082][C-0000000b] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-0000001a' [Jun 22 07:06:15] VERBOSE[17083][C-0000000b] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:06:15] VERBOSE[17083][C-0000000b] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-0000001a [Jun 22 07:06:15] VERBOSE[17090][C-0000000b] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:06:15] VERBOSE[17090][C-0000000b] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-0000001a [Jun 22 07:06:25] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:55162 [Jun 22 07:06:26] VERBOSE[13409] chan_sip.c: Reloading SIP [Jun 22 07:06:31] VERBOSE[13409] chan_sip.c: Reloading SIP [Jun 22 07:06:35] NOTICE[13409] chan_sip.c: Peer '3115' is now UNREACHABLE! Last qualify: 7 [Jun 22 07:06:35] VERBOSE[17300] pbx_spool.c: Attempting call on SIP/3115/9619928627 for application queue(support-installer1) (Retry 1) [Jun 22 07:06:35] NOTICE[17300] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jun 22 07:06:35] NOTICE[17300] pbx_spool.c: Queued call to SIP/3115/9619928627 expired without completion after 0 attempts [Jun 22 07:06:45] NOTICE[13409] chan_sip.c: Peer '3115' is now Reachable. (5ms / 2000ms) [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] res_rtp_asterisk.c: 0x7fd278061ee0 -- Strict RTP learning after remote address set to: 172.16.15.151:10866 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-0000001c", "02268388081") in new stack [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] app_verbose.c: 02268388081 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-0000001c", "number=8081") in new stack [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-0000001c", "8081") in new stack [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] app_verbose.c: 8081 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-0000001c", "1?kafka:") in new stack [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-0000001c", "** Channel SIP/02268388081-0000001c queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:06:46] WARNING[17303][C-0000000c] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592809606EPOCH'? [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-0000001c", "22062020-07:06:46-02268388081-customer-incoming.wav") in new stack [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-0000001c", "support-installer3") in new stack [Jun 22 07:06:46] VERBOSE[17304][C-0000000c] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-0000001c [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-0000001c' [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] app_queue.c: Called SIP/9010 [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:06:46] VERBOSE[13409][C-0000000c] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP learning after remote address set to: 172.16.14.104:41949 [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] app_queue.c: SIP/9010-0000001d answered SIP/02268388081-0000001c [Jun 22 07:06:46] NOTICE[17303][C-0000000c] app_queue.c: Delaying member connect for 2 seconds [Jun 22 07:06:46] VERBOSE[17303][C-0000000c] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP switching to RTP target address 172.16.14.104:41949 as source [Jun 22 07:06:48] VERBOSE[17303][C-0000000c] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 07:06:50] VERBOSE[17303][C-0000000c] file.c: Playing 'digits/4.ulaw' (language 'en') [Jun 22 07:06:50] VERBOSE[17303][C-0000000c] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 07:06:51] VERBOSE[17303][C-0000000c] res_musiconhold.c: Stopped music on hold on SIP/02268388081-0000001c [Jun 22 07:06:51] VERBOSE[17309][C-0000000c] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-0000001c [Jun 22 07:06:51] VERBOSE[17312][C-0000000c] bridge_channel.c: Channel SIP/9010-0000001d joined 'simple_bridge' basic-bridge <45b8408e-5225-472c-842a-fdab01151902> [Jun 22 07:06:51] VERBOSE[17303][C-0000000c] bridge_channel.c: Channel SIP/02268388081-0000001c joined 'simple_bridge' basic-bridge <45b8408e-5225-472c-842a-fdab01151902> [Jun 22 07:06:52] VERBOSE[17312][C-0000000c] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP learning complete - Locking on source address 172.16.14.104:41949 [Jun 22 07:06:52] VERBOSE[17303][C-0000000c] res_rtp_asterisk.c: 0x7fd278061ee0 -- Strict RTP switching to RTP target address 172.16.15.151:10866 as source [Jun 22 07:06:52] VERBOSE[17303][C-0000000c] res_rtp_asterisk.c: 0x7fd278061ee0 -- Strict RTP learning complete - Locking on source address 172.16.15.151:10866 [Jun 22 07:07:15] VERBOSE[17409] pbx_spool.c: Attempting call on SIP/3115/9619928627 for application queue(support-installer1) (Retry 1) [Jun 22 07:07:15] VERBOSE[17409] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:07:15] VERBOSE[17409] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:07:15] VERBOSE[17409] dial.c: Called 3115/9619928627 [Jun 22 07:07:15] WARNING[13409] chan_sip.c: Received response: "Forbidden" from '"9619928627 'I want to deposit money'" ;tag=as357b92b0' [Jun 22 07:07:15] NOTICE[17409] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jun 22 07:07:15] NOTICE[17409] pbx_spool.c: Queued call to SIP/3115/9619928627 expired without completion after 0 attempts [Jun 22 07:07:36] NOTICE[13409] chan_sip.c: Peer '9010' is now UNREACHABLE! Last qualify: 1 [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] bridge_channel.c: Channel SIP/02268388081-0000001c left 'simple_bridge' basic-bridge <45b8408e-5225-472c-842a-fdab01151902> [Jun 22 07:07:52] VERBOSE[17312][C-0000000c] bridge_channel.c: Channel SIP/9010-0000001d left 'simple_bridge' basic-bridge <45b8408e-5225-472c-842a-fdab01151902> [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-0000001c' [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-0000001c", "GLOBAL(call_end_time)=2020-06-22 07:07:52") in new stack [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:07:52' [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-0000001c", ""call Hangup details - - 7 - 31 - /var/spool/asterisk/monitor/1592809606.39.wav - 2020-06-22 07:06:46 - 2020-06-22 06:25:09 - 2020-06-22 07:07:52 - 2020-06-22 07:07:52 - "") in new stack [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-0000001c", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-0000001c", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-0000001c", "") in new stack [Jun 22 07:07:52] VERBOSE[17303][C-0000000c] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-0000001c' [Jun 22 07:07:52] VERBOSE[17304][C-0000000c] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:07:52] VERBOSE[17304][C-0000000c] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-0000001c [Jun 22 07:07:52] VERBOSE[17309][C-0000000c] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:07:52] VERBOSE[17309][C-0000000c] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-0000001c [Jun 22 07:07:58] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:33525 [Jun 22 07:07:58] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms) [Jun 22 07:08:20] VERBOSE[17805] pbx_spool.c: Attempting call on SIP/3115/9619928627 for application queue(support-installer1) (Retry 1) [Jun 22 07:08:20] VERBOSE[17805] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:08:20] VERBOSE[17805] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:08:20] VERBOSE[17805] dial.c: Called 3115/9619928627 [Jun 22 07:08:20] WARNING[13409] chan_sip.c: Received response: "Forbidden" from '"9619928627 'I want to deposit money'" ;tag=as3bc3a688' [Jun 22 07:08:20] NOTICE[17805] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jun 22 07:08:20] NOTICE[17805] pbx_spool.c: Queued call to SIP/3115/9619928627 expired without completion after 0 attempts [Jun 22 07:09:17] VERBOSE[17820] pbx_spool.c: Attempting call on SIP/3115/9619928627 for application queue(support-installer1) (Retry 1) [Jun 22 07:09:17] VERBOSE[17820] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:09:17] VERBOSE[17820] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:09:17] VERBOSE[17820] dial.c: Called 3115/9619928627 [Jun 22 07:09:17] WARNING[13409] chan_sip.c: Retransmission timeout reached on transmission 63017f3d0cb4e99320c673ef5078f950@172.16.15.151 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 448ms with no response [Jun 22 07:09:17] WARNING[13409] chan_sip.c: Hanging up call 63017f3d0cb4e99320c673ef5078f950@172.16.15.151 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Jun 22 07:09:17] NOTICE[17820] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jun 22 07:09:17] NOTICE[17820] pbx_spool.c: Queued call to SIP/3115/9619928627 expired without completion after 0 attempts [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] res_rtp_asterisk.c: 0x7fd27806f270 -- Strict RTP learning after remote address set to: 172.16.15.151:18630 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000021", "02268388081") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] app_verbose.c: 02268388081 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000021", "number=8081") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000021", "8081") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] app_verbose.c: 8081 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000021", "1?kafka:") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000021", "** Channel SIP/02268388081-00000021 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:14:31] WARNING[17887][C-0000000d] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592810071EPOCH'? [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000021", "22062020-07:14:31-02268388081-customer-incoming.wav") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000021", "support-installer3") in new stack [Jun 22 07:14:31] VERBOSE[17888][C-0000000d] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000021 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000021' [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] app_queue.c: Called SIP/9010 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000021 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000021' [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000021", "GLOBAL(call_end_time)=2020-06-22 07:14:31") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:14:31' [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000021", ""call Hangup details - - - 31 - /var/spool/asterisk/monitor/22062020-07:14:31-02268388081-customer-incoming.wav - 2020-06-22 07:14:31 - 2020-06-22 06:25:09 - 2020-06-22 07:14:31 - 2020-06-22 07:14:31 - "") in new stack [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:14:31] VERBOSE[13409][C-0000000d] res_rtp_asterisk.c: 0x7fd2b80094d0 -- Strict RTP learning after remote address set to: 172.16.14.104:59138 [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000021", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000021", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000021", "") in new stack [Jun 22 07:14:31] VERBOSE[17887][C-0000000d] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000021' [Jun 22 07:14:31] VERBOSE[17888][C-0000000d] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:14:31] VERBOSE[17888][C-0000000d] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000021 [Jun 22 07:15:10] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:47506 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] res_rtp_asterisk.c: 0x7fd2780727a0 -- Strict RTP learning after remote address set to: 172.16.15.151:11288 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000023", "02268388081") in new stack [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] app_verbose.c: 02268388081 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000023", "number=8081") in new stack [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000023", "8081") in new stack [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] app_verbose.c: 8081 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000023", "1?kafka:") in new stack [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000023", "** Channel SIP/02268388081-00000023 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:16:21] WARNING[18016][C-0000000e] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592810181EPOCH'? [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000023", "22062020-07:16:21-02268388081-customer-incoming.wav") in new stack [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000023", "support-installer3") in new stack [Jun 22 07:16:21] VERBOSE[18017][C-0000000e] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000023 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000023' [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] app_queue.c: Called SIP/9010 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:16:21] VERBOSE[13409][C-0000000e] res_rtp_asterisk.c: 0x7fd2c000e5f0 -- Strict RTP learning after remote address set to: 172.16.14.104:36158 [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] app_queue.c: SIP/9010-00000024 answered SIP/02268388081-00000023 [Jun 22 07:16:21] NOTICE[18016][C-0000000e] app_queue.c: Delaying member connect for 2 seconds [Jun 22 07:16:21] VERBOSE[18016][C-0000000e] res_rtp_asterisk.c: 0x7fd2c000e5f0 -- Strict RTP switching to RTP target address 172.16.14.104:36158 as source [Jun 22 07:16:23] VERBOSE[18016][C-0000000e] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 07:16:24] VERBOSE[18016][C-0000000e] file.c: Playing 'digits/3.ulaw' (language 'en') [Jun 22 07:16:24] VERBOSE[18016][C-0000000e] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 07:16:26] VERBOSE[18016][C-0000000e] res_rtp_asterisk.c: 0x7fd2c000e5f0 -- Strict RTP learning complete - Locking on source address 172.16.14.104:36158 [Jun 22 07:16:26] VERBOSE[18016][C-0000000e] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000023 [Jun 22 07:16:26] VERBOSE[18024][C-0000000e] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000023 [Jun 22 07:16:26] VERBOSE[18027][C-0000000e] bridge_channel.c: Channel SIP/9010-00000024 joined 'simple_bridge' basic-bridge [Jun 22 07:16:26] VERBOSE[18016][C-0000000e] bridge_channel.c: Channel SIP/02268388081-00000023 joined 'simple_bridge' basic-bridge [Jun 22 07:16:26] VERBOSE[18016][C-0000000e] res_rtp_asterisk.c: 0x7fd2780727a0 -- Strict RTP switching to RTP target address 172.16.15.151:11288 as source [Jun 22 07:16:26] VERBOSE[18016][C-0000000e] res_rtp_asterisk.c: 0x7fd2780727a0 -- Strict RTP learning complete - Locking on source address 172.16.15.151:11288 [Jun 22 07:17:09] NOTICE[13409] chan_sip.c: Peer '9010' is now UNREACHABLE! Last qualify: 1 [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] bridge_channel.c: Channel SIP/02268388081-00000023 left 'simple_bridge' basic-bridge [Jun 22 07:17:28] VERBOSE[18027][C-0000000e] bridge_channel.c: Channel SIP/9010-00000024 left 'simple_bridge' basic-bridge [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000023' [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000023", "GLOBAL(call_end_time)=2020-06-22 07:17:28") in new stack [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:17:28' [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000023", ""call Hangup details - - 8 - 38 - /var/spool/asterisk/monitor/1592810181.48.wav - 2020-06-22 07:16:21 - 2020-06-22 06:25:09 - 2020-06-22 07:17:28 - 2020-06-22 07:17:28 - "") in new stack [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000023", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000023", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000023", "") in new stack [Jun 22 07:17:28] VERBOSE[18016][C-0000000e] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000023' [Jun 22 07:17:28] VERBOSE[18024][C-0000000e] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:17:28] VERBOSE[18017][C-0000000e] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:17:28] VERBOSE[18017][C-0000000e] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000023 [Jun 22 07:17:28] VERBOSE[18024][C-0000000e] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000023 [Jun 22 07:17:29] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:47206 [Jun 22 07:17:29] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms) [Jun 22 07:19:18] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:35747 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] res_rtp_asterisk.c: 0x7fd278071230 -- Strict RTP learning after remote address set to: 172.16.15.151:18472 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:1] Verbose("SIP/02268388081-00000025", "02268388081") in new stack [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] app_verbose.c: 02268388081 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:2] Set("SIP/02268388081-00000025", "number=8081") in new stack [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:3] Verbose("SIP/02268388081-00000025", "8081") in new stack [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] app_verbose.c: 8081 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:4] GotoIf("SIP/02268388081-00000025", "1?kafka:") in new stack [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx_builtins.c: Goto (trunkinbound,02268388081,8) [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:8] NoOp("SIP/02268388081-00000025", "** Channel SIP/02268388081-00000025 queue sip hung up with Fax status with cause 0 **") in new stack [Jun 22 07:19:42] WARNING[18512][C-0000000f] pbx_functions.c: Can't find trailing parenthesis for function 'STRFTIME(1592810382EPOCH'? [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:9] MixMonitor("SIP/02268388081-00000025", "22062020-07:19:42-02268388081-customer-incoming.wav") in new stack [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] pbx.c: Executing [02268388081@trunkinbound:10] Queue("SIP/02268388081-00000025", "support-installer3") in new stack [Jun 22 07:19:42] VERBOSE[18513][C-0000000f] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000025 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/02268388081-00000025' [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] netsock2.c: Using SIP RTP TOS bits 184 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] netsock2.c: Using SIP RTP CoS mark 5 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] app_queue.c: Called SIP/9010 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] netsock2.c: Using UDPTL TOS bits 184 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] netsock2.c: Using UDPTL CoS mark 5 [Jun 22 07:19:42] VERBOSE[13409][C-0000000f] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP learning after remote address set to: 172.16.14.104:49156 [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] app_queue.c: SIP/9010-00000026 answered SIP/02268388081-00000025 [Jun 22 07:19:42] NOTICE[18512][C-0000000f] app_queue.c: Delaying member connect for 2 seconds [Jun 22 07:19:42] VERBOSE[18512][C-0000000f] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP switching to RTP target address 172.16.14.104:49156 as source [Jun 22 07:19:44] VERBOSE[18512][C-0000000f] file.c: Playing 'queue-reporthold.ulaw' (language 'en') [Jun 22 07:19:45] VERBOSE[18512][C-0000000f] file.c: Playing 'digits/3.ulaw' (language 'en') [Jun 22 07:19:46] VERBOSE[18512][C-0000000f] file.c: Playing 'queue-seconds.ulaw' (language 'en') [Jun 22 07:19:47] VERBOSE[18512][C-0000000f] res_rtp_asterisk.c: 0x7fd2c800e2e0 -- Strict RTP learning complete - Locking on source address 172.16.14.104:49156 [Jun 22 07:19:47] VERBOSE[18512][C-0000000f] res_musiconhold.c: Stopped music on hold on SIP/02268388081-00000025 [Jun 22 07:19:47] VERBOSE[18520][C-0000000f] app_mixmonitor.c: Begin MixMonitor Recording SIP/02268388081-00000025 [Jun 22 07:19:47] VERBOSE[18523][C-0000000f] bridge_channel.c: Channel SIP/9010-00000026 joined 'simple_bridge' basic-bridge <86ab488d-3479-4aea-9d38-edbd2231bdea> [Jun 22 07:19:47] VERBOSE[18512][C-0000000f] bridge_channel.c: Channel SIP/02268388081-00000025 joined 'simple_bridge' basic-bridge <86ab488d-3479-4aea-9d38-edbd2231bdea> [Jun 22 07:19:48] VERBOSE[18512][C-0000000f] res_rtp_asterisk.c: 0x7fd278071230 -- Strict RTP switching to RTP target address 172.16.15.151:18472 as source [Jun 22 07:19:48] VERBOSE[18512][C-0000000f] res_rtp_asterisk.c: 0x7fd278071230 -- Strict RTP learning complete - Locking on source address 172.16.15.151:18472 [Jun 22 07:20:22] NOTICE[13409] chan_sip.c: Peer '9010' is now UNREACHABLE! Last qualify: 1 [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] bridge_channel.c: Channel SIP/02268388081-00000025 left 'simple_bridge' basic-bridge <86ab488d-3479-4aea-9d38-edbd2231bdea> [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Spawn extension (trunkinbound, 02268388081, 10) exited non-zero on 'SIP/02268388081-00000025' [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Executing [h@trunkinbound:1] Set("SIP/02268388081-00000025", "GLOBAL(call_end_time)=2020-06-22 07:21:09") in new stack [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx_variables.c: Setting global variable 'call_end_time' to '2020-06-22 07:21:09' [Jun 22 07:21:09] VERBOSE[18523][C-0000000f] bridge_channel.c: Channel SIP/9010-00000026 left 'simple_bridge' basic-bridge <86ab488d-3479-4aea-9d38-edbd2231bdea> [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Executing [h@trunkinbound:2] NoOp("SIP/02268388081-00000025", ""call Hangup details - - 9 - 49 - /var/spool/asterisk/monitor/1592810382.51.wav - 2020-06-22 07:19:42 - 2020-06-22 06:25:09 - 2020-06-22 07:21:09 - 2020-06-22 07:21:09 - "") in new stack [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Executing [h@trunkinbound:3] Set("SIP/02268388081-00000025", "CRSLT={ "status": 200 }") in new stack [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Executing [h@trunkinbound:4] NoOp("SIP/02268388081-00000025", ""http response of pri call hangup is { "status": 200 }"") in new stack [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Executing [h@trunkinbound:5] Hangup("SIP/02268388081-00000025", "") in new stack [Jun 22 07:21:09] VERBOSE[18512][C-0000000f] pbx.c: Spawn extension (trunkinbound, h, 5) exited non-zero on 'SIP/02268388081-00000025' [Jun 22 07:21:09] VERBOSE[18520][C-0000000f] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:21:09] VERBOSE[18520][C-0000000f] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000025 [Jun 22 07:21:09] VERBOSE[18513][C-0000000f] app_mixmonitor.c: MixMonitor close filestream (mixed) [Jun 22 07:21:09] VERBOSE[18513][C-0000000f] app_mixmonitor.c: End MixMonitor Recording SIP/02268388081-00000025 [Jun 22 07:21:17] VERBOSE[13409] chan_sip.c: Registered SIP '9010' at 172.16.14.104:50049 [Jun 22 07:21:17] NOTICE[13409] chan_sip.c: Peer '9010' is now Reachable. (1ms / 2000ms)