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- SIP Debugging enabled
- == WebSocket connection from '192.168.88.174:49537' for protocol 'sip' accepted using version '13'
- <--- SIP read from WS:192.168.88.174:49537 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport
- From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
- To: "888"<sip:888@192.168.88.251>
- Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
- CSeq: 6286 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="888",realm="192.168.88.251",nonce="",uri="sip:192.168.88.251",response=""
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- Supported: path
- <------------->
- --- (14 headers 0 lines) ---
- <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
- To: "888"<sip:888@192.168.88.251>;tag=as7404acc2
- Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
- CSeq: 6286 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="09171086"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)
- <--- SIP read from WS:192.168.88.174:49537 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport
- From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
- To: "888"<sip:888@192.168.88.251>
- Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
- CSeq: 6287 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="888",realm="192.168.88.251",nonce="09171086",uri="sip:192.168.88.251",response="7392735b12a3d08cee2d5aa586a8f225",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- Supported: path
- <------------->
- --- (14 headers 0 lines) ---
- -- Registered SIP '888' at 192.168.88.174:49537
- <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
- To: "888"<sip:888@192.168.88.251>;tag=as7404acc2
- Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
- CSeq: 6287 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:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
- Date: Fri, 13 Feb 2015 04:15:38 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)
- <--- SIP read from WS:192.168.88.174:49537 --->
- INVITE sip:889@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>
- Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26632 INVITE
- Content-Type: application/sdp
- Content-Length: 1590
- 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=- 437287842917936700 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
- m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
- c=IN IP4 192.168.88.174
- a=rtcp:65003 IN IP4 192.168.88.174
- a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
- a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
- a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
- a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
- a=ice-ufrag:E3Z2rHZDSTilnu0z
- a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
- a=ice-options:google-ice
- a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
- 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:3257448580 cname:gVHxZ+t4jTqxgc7F
- a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
- a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
- a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
- <------------->
- --- (13 headers 39 lines) ---
- Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
- Found peer '888' for '888' from 192.168.88.174:49537
- <--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>;tag=as29afbf84
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26632 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="5a02570a"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)
- <--- SIP read from WS:192.168.88.174:49537 --->
- ACK sip:889@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>;tag=as29afbf84
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26632 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:192.168.88.174:49537 --->
- INVITE sip:889@192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>
- Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26633 INVITE
- Content-Type: application/sdp
- Content-Length: 1590
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:889@192.168.88.251",response="d328cce01e71bf1f662eae00a41ed1aa",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- v=0
- o=- 437287842917936700 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
- m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
- c=IN IP4 192.168.88.174
- a=rtcp:65003 IN IP4 192.168.88.174
- a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
- a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
- a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
- a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
- a=ice-ufrag:E3Z2rHZDSTilnu0z
- a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
- a=ice-options:google-ice
- a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
- 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:3257448580 cname:gVHxZ+t4jTqxgc7F
- a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
- a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
- a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
- <------------->
- --- (14 headers 39 lines) ---
- Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
- Found peer '888' for '888' from 192.168.88.174:49537
- == 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 192.168.88.174:65003
- Looking for 889 in default (domain 192.168.88.251)
- sip_route_dump: route/path hop: <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>
- <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26633 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:889@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- -- Executing [889@default:1] Dial("SIP/888-0000005f", "SIP/889") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 13566
- 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 192.168.88.187:49625:
- INVITE sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
- Max-Forwards: 70
- From: "888" <sip:888@192.168.88.251>;tag=as176937a0
- To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
- Contact: <sip:888@192.168.88.251:5060;transport=WS>
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 13.2.0
- Date: Fri, 13 Feb 2015 04:15:46 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 676
- v=0
- o=root 1187332515 1187332515 IN IP4 192.168.88.251
- s=Asterisk PBX 13.2.0
- c=IN IP4 192.168.88.251
- t=0 0
- m=audio 13566 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:2870a6d64720723e773aa0364395d0f4
- a=ice-pwd:1c240a294ddc471a2b5ba61158281a7d
- a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 13566 typ host
- a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 13567 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/889
- <--- SIP read from WS:192.168.88.187:49625 --->
- SIP/2.0 100 Trying (sent from the Transaction Layer)
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
- From: "888"<sip:888@192.168.88.251>;tag=as176937a0
- To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from WS:192.168.88.187:49625 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
- From: "888"<sip:888@192.168.88.251>;tag=as176937a0
- To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
- Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@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:889@df7jal23ls0d.invalid;transport=ws>
- -- SIP/889-00000060 is ringing
- <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>;tag=as4947f672
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26633 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:889@192.168.88.251:5060;transport=WS>
- Content-Length: 0
- <------------>
- <--- SIP read from WS:192.168.88.187:49625 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport
- From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
- To: "889"<sip:889@192.168.88.251>
- Contact: "889"<sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
- CSeq: 30884 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="889",realm="192.168.88.251",nonce="094b5901",uri="sip:192.168.88.251",response="a635f03607d21873ed8a4a74975b97cc",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport;received=192.168.88.187
- From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
- To: "889"<sip:889@192.168.88.251>;tag=as0ddfbb67
- Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
- CSeq: 30884 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="13fe51c6"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)
- <--- SIP read from WS:192.168.88.187:49625 --->
- REGISTER sip:192.168.88.251 SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport
- From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
- To: "889"<sip:889@192.168.88.251>
- Contact: "889"<sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
- Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
- CSeq: 30885 REGISTER
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="889",realm="192.168.88.251",nonce="13fe51c6",uri="sip:192.168.88.251",response="a1a4fccb04ffa5cef37a37267ce4cc4e",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport;received=192.168.88.187
- From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
- To: "889"<sip:889@192.168.88.251>;tag=as0ddfbb67
- Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
- CSeq: 30885 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:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
- Date: Fri, 13 Feb 2015 04:15:49 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)
- Really destroying SIP dialog 'd49645e7-984f-10b5-863d-bb5631bf9285' Method: REGISTER
- <--- SIP read from WS:192.168.88.187:49625 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
- From: "888"<sip:888@192.168.88.251>;tag=as176937a0
- To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
- Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 102 INVITE
- Content-Type: application/sdp
- Content-Length: 1171
- Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
- v=0
- o=- 7710960356793537000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=msid-semantic: WMS 85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
- m=audio 50026 UDP/TLS/RTP/SAVPF 0 8 101
- c=IN IP4 192.168.88.187
- a=rtcp:50027 IN IP4 192.168.88.187
- a=candidate:2577307183 1 udp 2122194687 192.168.88.187 50026 typ host generation 0
- a=candidate:2577307183 2 udp 2122194686 192.168.88.187 50027 typ host generation 0
- a=candidate:3609029343 1 tcp 1518214911 192.168.88.187 0 typ host tcptype active generation 0
- a=candidate:3609029343 2 tcp 1518214910 192.168.88.187 0 typ host tcptype active generation 0
- a=ice-ufrag:awyAyzrZujG0M9zq
- a=ice-pwd:z7uMvoajOVqtt3hHg+yAhqmj
- a=fingerprint:sha-256 D8:C4:BF:59:B9:A8:19:A0:4C:31:BA:92:F0:62:A0:3E:27:D4:90:9B:79:33:E3:B6:FC:E9:2A:EB:C3:D3:DF:E6
- 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:372692495 cname:pkv1+OHhG1pbR11N
- a=ssrc:372692495 msid:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO 4890e5af-7a2d-4d88-a39f-32216cbc0d0d
- a=ssrc:372692495 mslabel:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
- a=ssrc:372692495 label:4890e5af-7a2d-4d88-a39f-32216cbc0d0d
- <------------->
- --- (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 192.168.88.187:50026
- sip_route_dump: route/path hop: <sip:889@df7jal23ls0d.invalid;transport=ws>
- [Feb 13 06:15:59] ERROR[1055][C-00000031]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
- [Feb 13 06:15:59] WARNING[1055][C-00000031]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
- set_destination: Parsing <sip:889@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 192.168.88.187:49625:
- ACK sip:889@df7jal23ls0d.invalid;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3a7ff3e4
- Max-Forwards: 70
- From: "888" <sip:888@192.168.88.251>;tag=as176937a0
- To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
- Contact: <sip:888@192.168.88.251:5060;transport=WS>
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 13.2.0
- Content-Length: 0
- ---
- -- SIP/889-00000060 answered SIP/888-0000005f
- Audio is at 16190
- 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 192.168.88.174:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>;tag=as4947f672
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26633 INVITE
- Server: Asterisk PBX 13.2.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:889@192.168.88.251:5060;transport=WS>
- Content-Type: application/sdp
- Content-Length: 673
- v=0
- o=root 380639406 380639406 IN IP4 192.168.88.251
- s=Asterisk PBX 13.2.0
- c=IN IP4 192.168.88.251
- t=0 0
- m=audio 16190 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:5c8047ce65b257e40fd0093a148878ce
- a=ice-pwd:4988ef1c009a1c717d3e1c2444e686ff
- a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 16190 typ host
- a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 16191 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/888-0000005f joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
- -- Channel SIP/889-00000060 joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
- <--- SIP read from WS:192.168.88.174:49537 --->
- ACK sip:889@192.168.88.251:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKClVdt1n8PFKSu96LXuzf;rport
- From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- To: <sip:889@192.168.88.251>;tag=as4947f672
- Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 26633 ACK
- Content-Length: 0
- Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
- Max-Forwards: 70
- Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:889@192.168.88.251:5060;transport=WS",response="780d8ba6cec222db3fac9c9f8de84e87",algorithm=MD5
- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
- Organization: Doubango Telecom
- <------------->
- --- (13 headers 0 lines) ---
- > 0x7fd8389ed7e0 -- Probation passed - setting RTP source address to 192.168.88.187:50026
- <--- SIP read from WS:192.168.88.187:49625 --->
- BYE sip:888@192.168.88.251:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport
- From: <sip:889@df7jal23ls0d.invalid>;tag=7NfJJeWFCMBeSuhOFJ3T
- To: "888"<sip:888@192.168.88.251>;tag=as176937a0
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 43548 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 '63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060' in 32000 ms (Method: BYE)
- <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport;received=192.168.88.187
- From: <sip:889@df7jal23ls0d.invalid>;tag=7NfJJeWFCMBeSuhOFJ3T
- To: "888"<sip:888@192.168.88.251>;tag=as176937a0
- Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
- CSeq: 43548 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/889-00000060 left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
- -- Channel SIP/888-0000005f left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
- == Spawn extension (default, 889, 1) exited non-zero on 'SIP/888-0000005f'
- Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)
- set_destination: Parsing <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;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 192.168.88.174:5060:
- BYE sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
- Max-Forwards: 70
- From: <sip:889@192.168.88.251>;tag=as4947f672
- To: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 102 BYE
- User-Agent: Asterisk PBX 13.2.0
- Proxy-Authorization: Digest username="888", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="5a02570a", response="b05e0a15f788390a4b32dd0b4e0b3ec7"
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- <--- SIP read from WS:192.168.88.174:49537 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
- From: <sip:889@192.168.88.251>;tag=as4947f672
- To: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
- Contact: <sip:888@df7jal23ls0d.invalid;transport=ws>
- Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
- CSeq: 102 BYE
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- SIP Response message for INCOMING dialog BYE arrived
- Really destroying SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' Method: INVITE
- Really destroying SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' Method: REGISTER
- Really destroying SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' Method: REGISTER
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