Asterisk 13.2.0, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 13.2.0 currently running on ast13 (pid = 23284)  <--- SIP read from UDP:192.168.2.111:5065 ---> REGISTER sip:192.168.2.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.111:5065;branch=z9hG4bK1294384246 From: "6015" ;tag=1990737108 To: "6015" Call-ID: 191359003@192.168.2.111 CSeq: 1 REGISTER Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.72.0.30 Expires: 3600 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.2.111:5065 (no NAT) Sending to 192.168.2.111:5065 (no NAT) <--- Transmitting (no NAT) to 192.168.2.111:5065 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.2.111:5065;branch=z9hG4bK1294384246;received=192.168.2.111 From: "6015" ;tag=1990737108 To: "6015" ;tag=as7535cc19 Call-ID: 191359003@192.168.2.111 CSeq: 1 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Apr 30 14:32:09] NOTICE[23333]: chan_sip.c:27866 handle_request_register: Registration from '"6015" ' failed for '192.168.2.111:5065' - No matching peer found Scheduling destruction of SIP dialog '191359003@192.168.2.111' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '1741096100@192.168.2.111' Method: REGISTER  <--- SIP read from UDP:192.168.2.71:5060 ---> INVITE sip:6503@192.168.2.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac55434326 Max-Forwards: 70 From: ;tag=1c55422555 To: Call-ID: 554214513042015133211@192.168.2.71 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 600/v.6.40A.047.009 Content-Type: application/sdp Content-Length: 276 v=0 o=AudiocodesGW 55402047 55401743 IN IP4 192.168.2.71 s=Phone-Call c=IN IP4 192.168.2.71 t=0 0 m=audio 6510 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 13 lines) --- Sending to 192.168.2.71:5060 (no NAT) Sending to 192.168.2.71:5060 (no NAT) Using INVITE request as basis request - 554214513042015133211@192.168.2.71 Found peer 'Audiocodes' for '0004369918586292' from 192.168.2.71:5060 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.71:6510 Looking for 6503 in DLPN_DialPlan1 (domain 192.168.2.4) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 192.168.2.71:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac55434326;received=192.168.2.71 From: ;tag=1c55422555 To: Call-ID: 554214513042015133211@192.168.2.71 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 13826 Adding codec alaw to SDP Adding codec g729 to SDP Adding codec g723 to SDP Adding codec ulaw to SDP Adding codec gsm to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac55434326;received=192.168.2.71 From: ;tag=1c55422555 To: ;tag=as6c6c83e5 Call-ID: 554214513042015133211@192.168.2.71 CSeq: 1 INVITE Server: Asterisk PBX 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: 1314 v=0 o=root 802419269 802419269 IN IP4 192.168.2.4 s=Asterisk PBX 13.2.0 c=IN IP4 192.168.2.4 t=0 0 m=audio 13826 RTP/AVP 8 18 4 0 3 111 112 5 10 118 7 110 117 119 97 9 102 115 116 107 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=ice-ufrag:71359acc0de2079465fb7a4607fb4f41 a=ice-pwd:125512bc6d1bd43f5e4ed11e1818e3c8 a=candidate:Hc0a80204 1 UDP 2130706431 192.168.2.4 13826 typ host a=candidate:Sc0a80204 1 UDP 1694498815 192.168.2.4 13826 typ srflx raddr 192.168.2.4 rport 13826 a=candidate:Hc0a80204 2 UDP 2130706430 192.168.2.4 13827 typ host a=candidate:Sc0a80204 2 UDP 1694498814 192.168.2.4 13827 typ srflx raddr 192.168.2.4 rport 13827 a=sendrecv <------------>  <--- SIP read from UDP:192.168.2.71:5060 ---> ACK sip:6503@192.168.2.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac56231801 Max-Forwards: 70 From: ;tag=1c55422555 To: ;tag=as6c6c83e5 Call-ID: 554214513042015133211@192.168.2.71 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 600/v.6.40A.047.009 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '45728d8c217cf251189dfd0608bbb679@192.168.2.4:5060' Method: INVITE Really destroying SIP dialog '1c71261866ba462916ede7511341b05c@192.168.2.4:5060' Method: INVITE [Apr 30 14:32:16] ERROR[23454][C-00000002]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Apr 30 14:32:16] WARNING[23454][C-00000002]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid' Audio is at 19906 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.214:57825: INVITE sips:6033@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.2.4:5060;branch=z9hG4bK2caed11d;rport Max-Forwards: 70 From: "'Accounting'" ;tag=as44c0f112 To: Contact: Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 30 Apr 2015 12:32:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "'Accounting'" ;party=calling;privacy=off;screen=no Diversion: ;reason=unknown Content-Type: application/sdp Content-Length: 849 v=0 o=root 1870942552 1870942552 IN IP4 192.168.2.4 s=Asterisk PBX 13.2.0 c=IN IP4 192.168.2.4 t=0 0 m=audio 19906 RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:3e083bd37447a24c2fe7378a41c957fe a=ice-pwd:1e8debab6cfb6a902b6a91876e1ea2ab a=candidate:Hc0a80204 1 UDP 2130706431 192.168.2.4 19906 typ host a=candidate:Sc0a80204 1 UDP 1694498815 192.168.2.4 19906 typ srflx raddr 192.168.2.4 rport 19906 a=candidate:Hc0a80204 2 UDP 2130706430 192.168.2.4 19907 typ host a=candidate:Sc0a80204 2 UDP 1694498814 192.168.2.4 19907 typ srflx raddr 192.168.2.4 rport 19907 a=connection:new a=setup:actpass a=fingerprint:SHA-256 48:6E:F9:1B:95:3C:47:F2:0B:E0:35:A8:5B:B6:B6:A9:53:BF:1D:E1:7E:AF:9E:EA:CE:49:70:4D:B6:6E:CE:6E a=sendrecv ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK2caed11d From: "'Accounting'";tag=as44c0f112 To: Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK2caed11d From: "'Accounting'";tag=as44c0f112 To: ;tag=SleTbTk8Z3TO5TuLpc9i Contact: Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop:  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 603 Failed to get local SDP Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK2caed11d From: "'Accounting'";tag=as44c0f112 To: ;tag=SleTbTk8Z3TO5TuLpc9i Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 INVITE Content-Length: 0 Reason: SIP; cause=603; text="Failed to get local SDP" <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 192.168.1.214:57825: ACK sips:6033@df7jal23ls0d.invalid;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.2.4:5060;branch=z9hG4bK2caed11d;rport Max-Forwards: 70 From: "'Accounting'" ;tag=as44c0f112 To: ;tag=SleTbTk8Z3TO5TuLpc9i Contact: Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 481 Dialog/Transaction Does Not Exist Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK2caed11d From: "'Accounting'";tag=as44c0f112 To: ;tag=SleTbTk8Z3TO5TuLpc9i Call-ID: 3cd3b32e7d522598774322724634bc12@192.168.2.4:5060 CSeq: 102 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Apr 30 14:32:16] WARNING[23383][C-00000002]: chan_sip.c:24306 handle_response: Remote host can't match request ACK to call '3cd3b32e7d522598774322724634bc12@192.168.2.4:5060'. Giving up. Really destroying SIP dialog '3cd3b32e7d522598774322724634bc12@192.168.2.4:5060' Method: INVITE Really destroying SIP dialog '59d4ed250b0422ab121a34282c475e18@192.168.2.4:5060' Method: INVITE Really destroying SIP dialog '1ecd4ed17798b1e05f7cb9640f0579a7@192.168.2.4:5060' Method: INVITE [Apr 30 14:32:21] ERROR[23454][C-00000002]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Apr 30 14:32:21] WARNING[23454][C-00000002]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid' Audio is at 12920 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.214:57825: INVITE sips:6033@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.2.4:5060;branch=z9hG4bK7d2ca0ee;rport Max-Forwards: 70 From: "'Accounting'" ;tag=as0daec841 To: Contact: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 30 Apr 2015 12:32:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "'Accounting'" ;party=calling;privacy=off;screen=no Diversion: ;reason=unknown Content-Type: application/sdp Content-Length: 849 v=0 o=root 1284928532 1284928532 IN IP4 192.168.2.4 s=Asterisk PBX 13.2.0 c=IN IP4 192.168.2.4 t=0 0 m=audio 12920 RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:44a7158b31a7e91037fc5ece62811c7c a=ice-pwd:54d5f6b26a49516974e9472f7281247a a=candidate:Hc0a80204 1 UDP 2130706431 192.168.2.4 12920 typ host a=candidate:Sc0a80204 1 UDP 1694498815 192.168.2.4 12920 typ srflx raddr 192.168.2.4 rport 12920 a=candidate:Hc0a80204 2 UDP 2130706430 192.168.2.4 12921 typ host a=candidate:Sc0a80204 2 UDP 1694498814 192.168.2.4 12921 typ srflx raddr 192.168.2.4 rport 12921 a=connection:new a=setup:actpass a=fingerprint:SHA-256 48:6E:F9:1B:95:3C:47:F2:0B:E0:35:A8:5B:B6:B6:A9:53:BF:1D:E1:7E:AF:9E:EA:CE:49:70:4D:B6:6E:CE:6E a=sendrecv ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK7d2ca0ee From: "'Accounting'";tag=as0daec841 To: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK7d2ca0ee From: "'Accounting'";tag=as0daec841 To: ;tag=S5AUzaOLbL0oXK6Oe906 Contact: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: Really destroying SIP dialog '1b52df06-f0e4-ced1-2de3-0965772a3a7c' Method: REGISTER  <--- SIP read from UDP:192.168.5.217:2049 ---> REGISTER sip:ast13.youcon.hq:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.217:2049;branch=z9hG4bK-igxhv8xx0dzg;rport From: "6013 SIP" ;tag=r8ncq7aph1 To: "6013 SIP" Call-ID: 55195431dc14-q09rbd2rf5jd CSeq: 3157 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/8.7.3.25 Allow-Events: dialog X-Real-IP: 192.168.5.217 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.5.217:2049 (no NAT) Sending to 192.168.5.217:2049 (no NAT)  <--- Transmitting (no NAT) to 192.168.5.217:2049 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.5.217:2049;branch=z9hG4bK-igxhv8xx0dzg;received=192.168.5.217;rport=2049 From: "6013 SIP" ;tag=r8ncq7aph1 To: "6013 SIP" ;tag=as5a9624fe Call-ID: 55195431dc14-q09rbd2rf5jd CSeq: 3157 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ast13.youcon.hq", nonce="579a074f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '55195431dc14-q09rbd2rf5jd' in 32000 ms (Method: REGISTER)  <--- SIP read from UDP:192.168.5.217:2049 ---> REGISTER sip:ast13.youcon.hq:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.217:2049;branch=z9hG4bK-jc91tq28zz5z;rport From: "6013 SIP" ;tag=r8ncq7aph1 To: "6013 SIP" Call-ID: 55195431dc14-q09rbd2rf5jd CSeq: 3158 REGISTER Max-Forwards: 70 Contact: ;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/8.7.3.25 Allow-Events: dialog X-Real-IP: 192.168.5.217 Supported: path, gruu Authorization: Digest username="6013",realm="ast13.youcon.hq",nonce="579a074f",uri="sip:ast13.youcon.hq:5060",response="4d4f728671f5cb977e30c8583b56a086",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.5.217:2049 (no NAT)  <--- Transmitting (no NAT) to 192.168.5.217:2049 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.217:2049;branch=z9hG4bK-jc91tq28zz5z;received=192.168.5.217;rport=2049 From: "6013 SIP" ;tag=r8ncq7aph1 To: "6013 SIP" ;tag=as5a9624fe Call-ID: 55195431dc14-q09rbd2rf5jd CSeq: 3158 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Thu, 30 Apr 2015 12:32:29 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '55195431dc14-q09rbd2rf5jd' in 32000 ms (Method: REGISTER)  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK7d2ca0ee From: "'Accounting'";tag=as0daec841 To: ;tag=S5AUzaOLbL0oXK6Oe906 Contact: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 2224 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE v=0 o=- 399659900735964740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=msid-semantic: WMS awoAeDFBU5zItyzrU3OM9QZDXzfuvx4deZSW m=audio 56561 UDP/TLS/RTP/SAVPF 8 0 101 c=IN IP4 192.168.1.214 a=rtcp:56565 IN IP4 192.168.1.214 a=candidate:2268300305 1 udp 2122260223 192.168.1.214 56561 typ host generation 0 a=candidate:665244166 1 udp 2122194687 192.168.23.1 56562 typ host generation 0 a=candidate:4207418597 1 udp 2122129151 169.254.117.56 56563 typ host generation 0 a=candidate:3084250109 1 udp 2122063615 192.168.19.1 56564 typ host generation 0 a=candidate:2268300305 2 udp 2122260222 192.168.1.214 56565 typ host generation 0 a=candidate:665244166 2 udp 2122194686 192.168.23.1 56566 typ host generation 0 a=candidate:4207418597 2 udp 2122129150 169.254.117.56 56567 typ host generation 0 a=candidate:3084250109 2 udp 2122063614 192.168.19.1 56568 typ host generation 0 a=candidate:3383785697 1 tcp 1518280447 192.168.1.214 0 typ host tcptype active generation 0 a=candidate:1764044534 1 tcp 1518214911 192.168.23.1 0 typ host tcptype active generation 0 a=candidate:3024871445 1 tcp 1518149375 169.254.117.56 0 typ host tcptype active generation 0 a=candidate:4183168781 1 tcp 1518083839 192.168.19.1 0 typ host tcptype active generation 0 a=candidate:3383785697 2 tcp 1518280446 192.168.1.214 0 typ host tcptype active generation 0 a=candidate:1764044534 2 tcp 1518214910 192.168.23.1 0 typ host tcptype active generation 0 a=candidate:3024871445 2 tcp 1518149374 169.254.117.56 0 typ host tcptype active generation 0 a=candidate:4183168781 2 tcp 1518083838 192.168.19.1 0 typ host tcptype active generation 0 a=ice-ufrag:gyl8Ka197897+/u1 a=ice-pwd:xIwtOM0dyxk8dfGqx/e2oa7B a=fingerprint:sha-256 8C:0F:7E:83:2E:72:C2:79:83:60:F9:29:09:2D:EF:28:10:A6:88:1B:87:19:FC:47:9B:9E:A6:8C:D4:BA:87:EA a=setup:active a=mid:audio a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ssrc:2462885833 cname:TwticBjDf/oY0gU3 a=ssrc:2462885833 msid:awoAeDFBU5zItyzrU3OM9QZDXzfuvx4deZSW 66a508c7-fdb0-4c2f-b836-60a38b07a000 a=ssrc:2462885833 mslabel:awoAeDFBU5zItyzrU3OM9QZDXzfuvx4deZSW a=ssrc:2462885833 label:66a508c7-fdb0-4c2f-b836-60a38b07a000 <-------------> --- (10 headers 37 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.214:56561 sip_route_dump: route/path hop: Transmitting (NAT) to 192.168.1.214:57825: ACK sips:6033@df7jal23ls0d.invalid;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.2.4:5060;branch=z9hG4bK13c6ca25;rport Max-Forwards: 70 From: "'Accounting'" ;tag=as0daec841 To: ;tag=S5AUzaOLbL0oXK6Oe906 Contact: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.71:5060 Audio is at 13826 Adding codec alaw to SDP Adding codec g729 to SDP Adding codec g723 to SDP Adding codec ulaw to SDP Adding codec gsm to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.2.71:5060: INVITE sip:0004369918586292@192.168.2.71:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK2f0aef10 Max-Forwards: 70 From: ;tag=as6c6c83e5 To: ;tag=1c55422555 Contact: Call-ID: 554214513042015133211@192.168.2.71 CSeq: 102 INVITE User-Agent: Asterisk PBX Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "webrtc_testuser3" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 1314 v=0 o=root 802419269 802419270 IN IP4 192.168.2.4 s=Asterisk PBX 13.2.0 c=IN IP4 192.168.2.4 t=0 0 m=audio 13826 RTP/AVP 8 18 4 0 3 111 112 5 10 118 7 110 117 119 97 9 102 115 116 107 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=ice-ufrag:71359acc0de2079465fb7a4607fb4f41 a=ice-pwd:125512bc6d1bd43f5e4ed11e1818e3c8 a=candidate:Hc0a80204 1 UDP 2130706431 192.168.2.4 13826 typ host a=candidate:Sc0a80204 1 UDP 1694498815 192.168.2.4 13826 typ srflx raddr 192.168.2.4 rport 13826 a=candidate:Hc0a80204 2 UDP 2130706430 192.168.2.4 13827 typ host a=candidate:Sc0a80204 2 UDP 1694498814 192.168.2.4 13827 typ srflx raddr 192.168.2.4 rport 13827 a=sendrecv ---  <--- SIP read from UDP:192.168.2.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK2f0aef10 From: ;tag=as6c6c83e5 To: ;tag=1c55422555 Call-ID: 554214513042015133211@192.168.2.71 CSeq: 102 INVITE Contact: Supported: em,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Require: timer Session-Expires: 1800;refresher=uac Min-SE: 90 Server: Audiocodes-Sip-Gateway-Mediant 600/v.6.40A.047.009 Content-Type: application/sdp Content-Length: 173 v=0 o=AudiocodesGW 55402047 55401744 IN IP4 192.168.2.71 s=Phone-Call c=IN IP4 192.168.2.71 t=0 0 m=audio 6510 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <-------------> --- (15 headers 9 lines) --- Found RTP audio format 8 Found audio description format PCMA for ID 8 failed to extend from 64 to 98 Capabilities: us - (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.71:6510 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.71:5060 Transmitting (no NAT) to 192.168.2.71:5060: ACK sip:0004369918586292@192.168.2.71:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK40757467 Max-Forwards: 70 From: ;tag=as6c6c83e5 To: ;tag=1c55422555 Contact: Call-ID: 554214513042015133211@192.168.2.71 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ---  <--- SIP read from UDP:192.168.2.71:5060 ---> BYE sip:6503@192.168.2.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac483061231 Max-Forwards: 70 From: ;tag=1c55422555 To: ;tag=as6c6c83e5 Call-ID: 554214513042015133211@192.168.2.71 CSeq: 2 BYE Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 600/v.6.40A.047.009 Reason: Q.850 ;cause=16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.2.71:5060 (no NAT) Scheduling destruction of SIP dialog '554214513042015133211@192.168.2.71' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.2.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.71;branch=z9hG4bKac483061231;received=192.168.2.71 From: ;tag=1c55422555 To: ;tag=as6c6c83e5 Call-ID: 554214513042015133211@192.168.2.71 CSeq: 2 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.1.214:57825: BYE sips:6033@df7jal23ls0d.invalid;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.2.4:5060;branch=z9hG4bK639b10cd;rport Max-Forwards: 70 From: "'Accounting'" ;tag=as0daec841 To: ;tag=S5AUzaOLbL0oXK6Oe906 Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 103 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 ---  <--- SIP read from WS:192.168.1.214:57825 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.2.4:5060;rport=5060;branch=z9hG4bK639b10cd From: "'Accounting'";tag=as0daec841 To: ;tag=S5AUzaOLbL0oXK6Oe906 Contact: Call-ID: 73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060 CSeq: 103 BYE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '73c19b370f9d5d114ec3591908f2484c@192.168.2.4:5060' Method: INVITE  <--- SIP read from UDP:192.168.2.111:5065 ---> REGISTER sip:192.168.2.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.111:5065;branch=z9hG4bK1045100485 From: "6015" ;tag=1509282287 To: "6015" Call-ID: 484936592@192.168.2.111 CSeq: 1 REGISTER Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.72.0.30 Expires: 3600 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.2.111:5065 (no NAT) Sending to 192.168.2.111:5065 (no NAT)  <--- Transmitting (no NAT) to 192.168.2.111:5065 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.2.111:5065;branch=z9hG4bK1045100485;received=192.168.2.111 From: "6015" ;tag=1509282287 To: "6015" ;tag=as1d404dd5 Call-ID: 484936592@192.168.2.111 CSeq: 1 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Apr 30 14:32:39] NOTICE[23333]: chan_sip.c:27866 handle_request_register: Registration from '"6015" ' failed for '192.168.2.111:5065' - No matching peer found Scheduling destruction of SIP dialog '484936592@192.168.2.111' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '191359003@192.168.2.111' Method: REGISTER