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- == Using SIP RTP CoS mark 5
- -- Executing [893@default:1] Dial("SIP/892-0000009e", "SIP/893") in new stack
- == Using SIP RTP CoS mark 5
- -- Called SIP/893
- -- SIP/893-0000009f is ringing
- [Feb 18 04:05:19] ERROR[5857][C-0000006d]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
- [Feb 18 04:05:19] WARNING[5857][C-0000006d]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
- -- SIP/893-0000009f answered SIP/892-0000009e
- -- Channel SIP/892-0000009e joined 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
- -- Channel SIP/893-0000009f joined 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
- > 0x7fe19d3a6080 -- Probation passed - setting RTP source address to 192.168.88.189:51710
- > 0x7fe19c49f1c0 -- Probation passed - setting RTP source address to 192.168.88.188:60363
- -- Channel SIP/893-0000009f left 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
- -- Channel SIP/892-0000009e left 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
- == Spawn extension (default, 893, 1) exited non-zero on 'SIP/892-0000009e'
- ast*CLI> sip set debug on
- SIP Debugging enabled
- <--- SIP read from WS:192.168.88.188:49766 --->
- INVITE sip:893@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>
- Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65211 INVITE
- Content-Type: application/sdp
- Content-Length: 1593
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- v=0
- o=- 9013918505325537000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
- m=audio 60364 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
- c=IN IP4 192.168.88.188
- a=rtcp:60364 IN IP4 192.168.88.188
- a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60364 typ host generation 0
- a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60364 typ host generation 0
- a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
- a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
- a=ice-ufrag:8mZ6nJPeh4HNMJ8u
- a=ice-pwd:RGdhONKCR6yEsEs8H/hiQx0U
- a=ice-options:google-ice
- a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
- a=setup:actpass
- a=mid:audio
- a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
- a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
- a=sendrecv
- a=rtcp-mux
- a=rtpmap:111 opus/48000/2
- a=fmtp:111 minptime=10
- a=rtpmap:103 ISAC/16000
- a=rtpmap:104 ISAC/32000
- a=rtpmap:9 G722/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:106 CN/32000
- a=rtpmap:105 CN/16000
- a=rtpmap:13 CN/8000
- a=rtpmap:126 telephone-event/8000
- a=maxptime:60
- a=ssrc:2479575733 cname:URhPGkQlkeqtQHWw
- a=ssrc:2479575733 msid:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE 64ee14c7-d87c-431d-8ed4-713c3117a869
- a=ssrc:2479575733 mslabel:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
- a=ssrc:2479575733 label:64ee14c7-d87c-431d-8ed4-713c3117a869
- <------------->
- --- (13 headers 39 lines) ---
- Using INVITE request as basis request - 7252dd20-3201-26b9-e94f-1754d3b61c84
- Found peer '892' for '892' from 192.168.88.188:49766
- <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport;received=192.168.88.188
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>;tag=as43bcc41a
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65211 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="1cded12e"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' in 32000 ms (Method: INVITE)
- <--- SIP read from WS:192.168.88.188:49766 --->
- ACK sip:893@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>;tag=as43bcc41a
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65211 ACK
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- <------------->
- --- (9 headers 0 lines) ---
- <--- SIP read from WS:192.168.88.188:49766 --->
- INVITE sip:893@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>
- Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65212 INVITE
- Content-Type: application/sdp
- Content-Length: 1593
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="892",realm="192.168.88.251",nonce="1cded12e",uri="sip:893@192.168.88.251",response="47dd36205a3c74b2fd4b21a1521dac3f",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- v=0
- o=- 9013918505325537000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
- m=audio 60364 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
- c=IN IP4 192.168.88.188
- a=rtcp:60364 IN IP4 192.168.88.188
- a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60364 typ host generation 0
- a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60364 typ host generation 0
- a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
- a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
- a=ice-ufrag:8mZ6nJPeh4HNMJ8u
- a=ice-pwd:RGdhONKCR6yEsEs8H/hiQx0U
- a=ice-options:google-ice
- a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
- a=setup:actpass
- a=mid:audio
- a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
- a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
- a=sendrecv
- a=rtcp-mux
- a=rtpmap:111 opus/48000/2
- a=fmtp:111 minptime=10
- a=rtpmap:103 ISAC/16000
- a=rtpmap:104 ISAC/32000
- a=rtpmap:9 G722/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:106 CN/32000
- a=rtpmap:105 CN/16000
- a=rtpmap:13 CN/8000
- a=rtpmap:126 telephone-event/8000
- a=maxptime:60
- a=ssrc:2479575733 cname:URhPGkQlkeqtQHWw
- a=ssrc:2479575733 msid:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE 64ee14c7-d87c-431d-8ed4-713c3117a869
- a=ssrc:2479575733 mslabel:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
- a=ssrc:2479575733 label:64ee14c7-d87c-431d-8ed4-713c3117a869
- <------------->
- --- (14 headers 39 lines) ---
- Using INVITE request as basis request - 7252dd20-3201-26b9-e94f-1754d3b61c84
- Found peer '892' for '892' from 192.168.88.188:49766
- == Using SIP RTP CoS mark 5
- Found RTP audio format 111
- Found RTP audio format 103
- Found RTP audio format 104
- Found RTP audio format 9
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 106
- Found RTP audio format 105
- Found RTP audio format 13
- Found RTP audio format 126
- Found audio description format opus for ID 111
- Found unknown media description format ISAC for ID 103
- Found unknown media description format ISAC for ID 104
- Found audio description format G722 for ID 9
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found unknown media description format CN for ID 106
- Found unknown media description format CN for ID 105
- Found audio description format CN for ID 13
- Found audio description format telephone-event for ID 126
- Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 192.168.88.188:60364
- Looking for 893 in default (domain 192.168.88.251)
- sip_route_dump: route/path hop: <sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
- <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65212 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:893@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- -- Executing [893@default:1] Dial("SIP/892-000000a0", "SIP/893") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 16278
- Adding codec ulaw to SDP
- Adding codec alaw to SDP
- Adding codec gsm to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 192.168.88.189:54237:
- INVITE sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
- Max-Forwards: 70
- From: "892" <sip:892@192.168.88.251>;tag=as3d7652ce
- To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
- Contact: <sip:892@192.168.88.251:5060;transport=WS>
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 13.2.0
- Date: Wed, 18 Feb 2015 02:05:35 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 674
- v=0
- o=root 250873218 250873218 IN IP4 192.168.88.251
- s=Asterisk PBX 13.2.0
- c=IN IP4 192.168.88.251
- t=0 0
- m=audio 16278 RTP/SAVPF 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=maxptime:150
- a=ice-ufrag:7ae4574e11e0910e742f5f4c4503d85e
- a=ice-pwd:0ba5d6ad73334552538a52503bc1da69
- a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 16278 typ host
- a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 16279 typ host
- a=connection:new
- a=setup:actpass
- a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
- a=sendrecv
- ---
- -- Called SIP/893
- <--- SIP read from WS:192.168.88.189:54237 --->
- SIP/2.0 100 Trying (sent from the Transaction Layer)
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
- From: "892"<sip:892@192.168.88.251>;tag=as3d7652ce
- To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from WS:192.168.88.189:54237 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
- From: "892"<sip:892@192.168.88.251>;tag=as3d7652ce
- To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
- Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- 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:893@df7jal23ls0d.invalid;transport=ws>
- -- SIP/893-000000a1 is ringing
- <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>;tag=as1a240a25
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65212 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:893@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- <--- SIP read from WS:192.168.88.189:54237 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhXsTmRcEND87oBsyUPTp9l2qrDJsKw2x;rport
- From: "893"<sip:893@192.168.88.251>;tag=LH6X6SThWalMGcJtquUq
- To: "893"<sip:893@192.168.88.251>
- Contact: "893"<sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
- CSeq: 27463 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="893",realm="192.168.88.251",nonce="3862f2d8",uri="sip:192.168.88.251",response="87ce2c4b8f87482ab0754ece3dd40052",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhXsTmRcEND87oBsyUPTp9l2qrDJsKw2x;rport;received=192.168.88.189
- From: "893"<sip:893@192.168.88.251>;tag=LH6X6SThWalMGcJtquUq
- To: "893"<sip:893@192.168.88.251>;tag=as2608aa5c
- Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
- CSeq: 27463 REGISTER
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="1d777b46"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc' in 32000 ms (Method: REGISTER)
- <--- SIP read from WS:192.168.88.189:54237 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOCZLWAod2T2lWC6yOUiCFTK3uAAQQvsd;rport
- From: "893"<sip:893@192.168.88.251>;tag=LH6X6SThWalMGcJtquUq
- To: "893"<sip:893@192.168.88.251>
- Contact: "893"<sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
- CSeq: 27464 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="893",realm="192.168.88.251",nonce="1d777b46",uri="sip:192.168.88.251",response="83f9e3fef5a8d6ca80eed28e40f86fca",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOCZLWAod2T2lWC6yOUiCFTK3uAAQQvsd;rport;received=192.168.88.189
- From: "893"<sip:893@192.168.88.251>;tag=LH6X6SThWalMGcJtquUq
- To: "893"<sip:893@192.168.88.251>;tag=as2608aa5c
- Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
- CSeq: 27464 REGISTER
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Expires: 200
- Contact: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
- Date: Wed, 18 Feb 2015 02:05:35 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc' in 32000 ms (Method: REGISTER)
- <--- SIP read from WS:192.168.88.189:54237 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
- From: "892"<sip:892@192.168.88.251>;tag=as3d7652ce
- To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
- Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 102 INVITE
- Content-Type: application/sdp
- Content-Length: 1173
- Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
- v=0
- o=- 6851900569207422000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=msid-semantic: WMS 5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa
- m=audio 51712 UDP/TLS/RTP/SAVPF 0 8 101
- c=IN IP4 192.168.88.189
- a=rtcp:51713 IN IP4 192.168.88.189
- a=candidate:2034360604 1 udp 2122194687 192.168.88.189 51712 typ host generation 0
- a=candidate:2034360604 2 udp 2122194686 192.168.88.189 51713 typ host generation 0
- a=candidate:935468524 1 tcp 1518214911 192.168.88.189 0 typ host tcptype active generation 0
- a=candidate:935468524 2 tcp 1518214910 192.168.88.189 0 typ host tcptype active generation 0
- a=ice-ufrag:17Hg6cuOE6d2IqDO
- a=ice-pwd:bguLRJ63C1EopK+2L82QoRMh
- a=fingerprint:sha-256 33:7A:5F:05:75:AB:62:A6:20:78:D5:F7:EF:BB:BB:3A:0A:D2:89:0F:DA:4D:22:50:8C:E7:90:A4:51:34:F8:61
- a=setup:active
- a=mid:audio
- a=sendrecv
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=ssrc:3629716700 cname:YYEK0OxOe3UE1N+f
- a=ssrc:3629716700 msid:5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa 0187d8f9-44c3-4140-8802-7381e4e403e8
- a=ssrc:3629716700 mslabel:5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa
- a=ssrc:3629716700 label:0187d8f9-44c3-4140-8802-7381e4e403e8
- <------------->
- --- (10 headers 25 lines) ---
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found audio description format telephone-event for ID 101
- Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 192.168.88.189:51712
- sip_route_dump: route/path hop: <sip:893@df7jal23ls0d.invalid;transport=ws>
- [Feb 18 04:05:39] ERROR[5857][C-0000006e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
- [Feb 18 04:05:39] WARNING[5857][C-0000006e]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
- set_destination: Parsing <sip:893@df7jal23ls0d.invalid;transport=ws> for address/port to send to
- set_destination: URI is for WebSocket, we can't set destination
- Transmitting (no NAT) to 192.168.88.189:54237:
- ACK sip:893@df7jal23ls0d.invalid;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3b200041
- Max-Forwards: 70
- From: "892" <sip:892@192.168.88.251>;tag=as3d7652ce
- To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
- Contact: <sip:892@192.168.88.251:5060;transport=WS>
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 13.2.0
- Content-Length: 0
- ---
- -- SIP/893-000000a1 answered SIP/892-000000a0
- Audio is at 13722
- Adding codec ulaw to SDP
- Adding codec alaw to SDP
- Adding codec gsm to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>;tag=as1a240a25
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65212 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:893@192.168.88.251:5060;transport=WS>
- Content-Type: application/sdp
- Content-Length: 673
- v=0
- o=root 382034728 382034728 IN IP4 192.168.88.251
- s=Asterisk PBX 13.2.0
- c=IN IP4 192.168.88.251
- t=0 0
- m=audio 13722 RTP/SAVPF 0 8 3 126
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:126 telephone-event/8000
- a=fmtp:126 0-16
- a=maxptime:150
- a=ice-ufrag:5a9b3168230d32be01020e2b48e838b2
- a=ice-pwd:31ddd4366598156e7885fd1d1932870f
- a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 13722 typ host
- a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 13723 typ host
- a=connection:new
- a=setup:active
- a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
- a=sendrecv
- <------------>
- -- Channel SIP/892-000000a0 joined 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
- -- Channel SIP/893-000000a1 joined 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
- <--- SIP read from WS:192.168.88.188:49766 --->
- ACK sip:893@192.168.88.251:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKl6PTQTDyGPWBoThqSJ15;rport
- From: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- To: <sip:893@192.168.88.251>;tag=as1a240a25
- Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 65212 ACK
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="892",realm="192.168.88.251",nonce="1cded12e",uri="sip:893@192.168.88.251:5060;transport=WS",response="55ff9d4edd610e4c776a49ae4c3b79b0",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- > 0x7fe19d38c3e0 -- Probation passed - setting RTP source address to 192.168.88.189:51712
- > 0x7fe19c49f1c0 -- Probation passed - setting RTP source address to 192.168.88.188:60364
- Really destroying SIP dialog 'dc8a463e-01f5-927f-6d0f-31c24e5c091e' Method: REGISTER
- <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS 192.0.2.157;branch=z9hG4bK4472428;received=192.168.88.182
- From: <sip:889@192.168.88.251>;tag=grberh41iu
- To: <sip:888@192.168.88.251>;tag=as35a3c936
- Call-ID: 61r71uq6tdlnd58mga7l
- CSeq: 5841 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:888@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- <--- SIP read from WS:192.168.88.189:54237 --->
- BYE sip:892@192.168.88.251:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxMk3DoB6czCTDkP8KmlAmfpIEEOCPnGp;rport
- From: <sip:893@df7jal23ls0d.invalid>;tag=i1J8u8fbSqqmTOnOGRim
- To: "892"<sip:892@192.168.88.251>;tag=as3d7652ce
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 2151 BYE
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Accept-Contact: *;+g.oma.sip-im
- Accept-Contact: *;language="en,fr"
- Accept-Contact: *;+g.oma.sip-im
- Accept-Contact: *;language="en,fr"
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (15 headers 0 lines) ---
- Scheduling destruction of SIP dialog '4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060' in 32000 ms (Method: BYE)
- <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxMk3DoB6czCTDkP8KmlAmfpIEEOCPnGp;rport;received=192.168.88.189
- From: <sip:893@df7jal23ls0d.invalid>;tag=i1J8u8fbSqqmTOnOGRim
- To: "892"<sip:892@192.168.88.251>;tag=as3d7652ce
- Call-ID: 4664d9454befaa332eb95a227b3fdbe4@192.168.88.251:5060
- CSeq: 2151 BYE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- -- Channel SIP/893-000000a1 left 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
- -- Channel SIP/892-000000a0 left 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
- == Spawn extension (default, 893, 1) exited non-zero on 'SIP/892-000000a0'
- Scheduling destruction of SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' in 32000 ms (Method: INVITE)
- set_destination: Parsing <sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
- set_destination: URI is for WebSocket, we can't set destination
- Reliably Transmitting (no NAT) to 192.168.88.188:5060:
- BYE sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK7ddb39ef
- Max-Forwards: 70
- From: <sip:893@192.168.88.251>;tag=as1a240a25
- To: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 102 BYE
- User-Agent: Asterisk PBX 13.2.0
- Proxy-Authorization: Digest username="889", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="1cded12e", response="06633f736457f7e9fcfe5d5359283a01"
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- <--- SIP read from WS:192.168.88.188:49766 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK7ddb39ef
- From: <sip:893@192.168.88.251>;tag=as1a240a25
- To: "892"<sip:892@192.168.88.251>;tag=WuTzQfG02C9oc1KGlJqA
- Contact: <sip:892@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
- CSeq: 102 BYE
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- SIP Response message for INCOMING dialog BYE arrived
- <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS 192.0.2.97;branch=z9hG4bK6600227;received=192.168.88.182
- From: <sip:889@192.168.88.251>;tag=ev8b7t1gfj
- To: <sip:888@192.168.88.251>;tag=as6bbeef1d
- Call-ID: s9iigcbrqmig8r9as7e7
- CSeq: 3015 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:888@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- Really destroying SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' Method: INVITE
- Really destroying SIP dialog '157b5b4a0b2317a60c4e35a80af2714c@192.168.88.251:5060' Method: BYE
- ast*CLI> sip set debug off
- SIP Debugging Disabled
- ast*CLI>
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