sip_route_dump: route/path hop: -- SIP/102-00000001 is ringing <--- Transmitting (NAT) to 103.249.82.14:58692 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9vRI77uL1ZS9L46F0Io7PsRCsEueH Hfn;received=103.249.82.14;rport=58692 From: "103";tag=OTpzbEAzuq0JR5MB1zUB To: ;tag=as21dd5dd9 Call-ID: 6a82b998-426d-c7e3-56bc-92f0899848f3 CSeq: 33108 INVITE Server: Asterisk PBX 15.5.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Retransmitting #2 (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK5fa6877d;rport Max-Forwards: 70 From: "asterisk" ;tag=as3f669e42 To: Contact: Call-ID: 480658f41dd011200dc0e5e942c5f31e@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK5fa6877d;rport Max-Forwards: 70 From: "asterisk" ;tag=as3f669e42 To: Contact: Call-ID: 480658f41dd011200dc0e5e942c5f31e@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK5fa6877d;rport Max-Forwards: 70 From: "asterisk" ;tag=as3f669e42 To: Contact: Call-ID: 480658f41dd011200dc0e5e942c5f31e@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '480658f41dd011200dc0e5e942c5f31e@13.233.25.20:5060 ' Method: OPTIONS <--- SIP read from WS:103.249.82.14:60126 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 13.233.25.20;rport;branch=z9hG4bK1c9902a4 From: "103";tag=as70fb4b56 To: ;tag=sqzCZuiL6 7L8LL7meWwt Contact: Call-ID: 11cf9af63d8eb739547614f231290a7f@13.233.25.20:0 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 783 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE v=0 o=mozilla...THIS_IS_SDPARTA-47.0 8183206358807532000 0 IN IP4 127.0.0.1 s=Doubango Telecom - firefox t=0 0 a=sendrecv a=fingerprint:sha-256 1B:37:A4:4C:EC:92:C5:44:48:F7:70:0E:3E:A7:72:3A:3B:19:2D:B A:80:2D:C2:4F:36:55:70:EE:70:0D:56:D1 a=ice-options:trickle a=msid-semantic:WMS * m=audio 48380 UDP/TLS/RTP/SAVPF 0 c=IN IP4 10.0.0.159 a=candidate:0 1 UDP 2122252543 10.0.0.159 48380 typ host a=candidate:0 2 UDP 2122252542 10.0.0.159 39620 typ host a=sendrecv a=end-of-candidates a=ice-pwd:c61e4ec11cbbe46643da338a5ec15330 a=ice-ufrag:c89d7306 a=msid:{5de83027-14ba-4259-b4b5-ad83353df88d} {16001f63-1ea3-4a68-a185-6fd0d6a65 636} a=rtcp:39620 IN IP4 10.0.0.159 a=rtpmap:0 PCMU/8000 a=setup:active a=ssrc:2836496812 cname:{acb3d872-fbf1-49bc-9b2c-73c59f740f0f} <-------------> --- (10 headers 21 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothin g), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing) , combined - 0x0 (nothing) Peer audio RTP is at port 10.0.0.159:48380 sip_route_dump: route/path hop: Transmitting (NAT) to 103.249.82.14:60126: ACK sip:102@df7jal23ls0d.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 13.233.25.20:0;branch=z9hG4bK7faf5049;rport Max-Forwards: 70 From: "103" ;tag=as70fb4b56 To: ;tag=sqzCZuiL6 7L8LL7meWwt Contact: Call-ID: 11cf9af63d8eb739547614f231290a7f@13.233.25.20:0 CSeq: 102 ACK User-Agent: Asterisk PBX 15.5.0 Content-Length: 0 --- -- SIP/102-00000001 answered SIP/103-00000000 Audio is at 53230 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 103.249.82.14:58692 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9vRI77uL1ZS9L46F0Io7PsRCsEueH Hfn;received=103.249.82.14;rport=58692 From: "103";tag=OTpzbEAzuq0JR5MB1zUB To: ;tag=as21dd5dd9 Call-ID: 6a82b998-426d-c7e3-56bc-92f0899848f3 CSeq: 33108 INVITE Server: Asterisk PBX 15.5.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 836 v=0 o=root 67408092 67408092 IN IP4 13.233.25.20 s=Asterisk PBX 15.5.0 c=IN IP4 13.233.25.20 t=0 0 m=audio 53230 RTP/SAVPF 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:11b4066e591505b005e55bb15558d2c0 a=ice-pwd:1c775a8c44d04d8967daec9d0b77baf4 a=candidate:Ha00012c 1 UDP 2130706431 10.0.1.44 53230 typ host a=candidate:Sde91914 1 UDP 1694498815 13.233.25.20 53230 typ srflx raddr 10.0.1. 44 rport 53230 a=candidate:Ha00012c 2 UDP 2130706430 10.0.1.44 53231 typ host a=candidate:Sde91914 2 UDP 1694498814 13.233.25.20 53231 typ srflx raddr 10.0.1. 44 rport 53231 a=connection:new a=setup:active a=fingerprint:SHA-256 77:C7:53:FE:53:36:B1:10:FF:F1:2E:91:A7:E4:CE:81:BF:AA:27:8 9:CD:2B:89:1A:D3:44:BB:D8:7A:8A:85:51 a=sendrecv <------------> -- Channel SIP/102-00000001 joined 'simple_bridge' basic-bridge -- Channel SIP/103-00000000 joined 'simple_bridge' basic-bridge <--- SIP read from WS:103.249.82.14:58692 ---> ACK sip:102@13.233.25.20;transport=ws SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhMmozkN7BkU9bZStRrG6;rport From: "103";tag=OTpzbEAzuq0JR5MB1zUB To: ;tag=as21dd5dd9 Contact: "103";+g.oma.sip-im;language="en,fr" Call-ID: 6a82b998-426d-c7e3-56bc-92f0899848f3 CSeq: 33108 ACK Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="103",realm="13.233.25.20",nonce="2dca46a8",uri=" sip:102@13.233.25.20;transport=ws",response="ea2c65a81e3a79e05dbd447a7dbed2c0",a lgorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom <-------------> --- (13 headers 0 lines) --- Reliably Transmitting (NAT) to 103.249.82.14:58692: OPTIONS sip:103@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 13.233.25.20:0;branch=z9hG4bK6472591a;rport Max-Forwards: 70 From: "asterisk" ;tag=as3dd3e24e To: Contact: Call-ID: 615bc57a43943db91d632308252e9be5@13.233.25.20:0 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from WS:103.249.82.14:58692 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 13.233.25.20;rport;branch=z9hG4bK6472591a From: "asterisk";tag=as3dd3e24e To: Call-ID: 615bc57a43943db91d632308252e9be5@13.233.25.20:0 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '615bc57a43943db91d632308252e9be5@13.233.25.20:0' M ethod: OPTIONS Reliably Transmitting (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK645cf650;rport Max-Forwards: 70 From: "asterisk" ;tag=as649f80a4 To: Contact: Call-ID: 6afbd44e408e4b13138ea4bc732bc725@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK645cf650;rport Max-Forwards: 70 From: "asterisk" ;tag=as649f80a4 To: Contact: Call-ID: 6afbd44e408e4b13138ea4bc732bc725@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (NAT) to 103.249.82.14:60098: OPTIONS sip:104@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/UDP 13.233.25.20:5060;branch=z9hG4bK645cf650;rport Max-Forwards: 70 From: "asterisk" ;tag=as649f80a4 To: Contact: Call-ID: 6afbd44e408e4b13138ea4bc732bc725@13.233.25.20:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from WS:103.249.82.14:60126 ---> REGISTER sip:13.233.25.20 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKq1FDXUK7roqJTtfd4a8F4oIh8kifM 0fA;rport From: "102";tag=4JAsJ61TZHUcDtMmdLq7 To: "102" Contact: "102";exp ires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 7c056035-3ab3-0c2e-b62a-e1134a7091b7 CSeq: 2830 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="102",realm="13.233.25.20",nonce="58fcaeff",uri=" sip:13.233.25.20",response="07901d6b815baa7acff4f8b15ac7e8f2",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom <-------------> --- (13 headers 0 lines) --- <--- Transmitting (NAT) to 103.249.82.14:60126 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKq1FDXUK7roqJTtfd4a8F4oIh8kifM 0fA;received=103.249.82.14;rport=60126 From: "102";tag=4JAsJ61TZHUcDtMmdLq7 To: "102";tag=as267fce92 Call-ID: 7c056035-3ab3-0c2e-b62a-e1134a7091b7 CSeq: 2830 REGISTER Server: Asterisk PBX 15.5.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="13.233.25.20", nonce="7982fd1d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7c056035-3ab3-0c2e-b62a-e1134a7091b7' in 3 2000 ms (Method: REGISTER) <--- SIP read from WS:103.249.82.14:60126 ---> REGISTER sip:13.233.25.20 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKm7zUleR6kamguXm1ky4t4UQVxPXc7 QIc;rport From: "102";tag=4JAsJ61TZHUcDtMmdLq7 To: "102" Contact: "102";exp ires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 7c056035-3ab3-0c2e-b62a-e1134a7091b7 CSeq: 2831 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="102",realm="13.233.25.20",nonce="7982fd1d",uri=" sip:13.233.25.20",response="c23e9a3693552924754bf977fd5b3778",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom <-------------> --- (13 headers 0 lines) --- Reliably Transmitting (NAT) to 103.249.82.14:60126: OPTIONS sip:102@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 13.233.25.20:0;branch=z9hG4bK66dce8f1;rport Max-Forwards: 70 From: "asterisk" ;tag=as51fbe0d5 To: Contact: Call-ID: 2acd60d0795f2b33398aba173201e1a7@13.233.25.20:0 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 15.5.0 Date: Mon, 27 Aug 2018 11:05:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE Supported: replaces, timer Content-Length: 0 ---