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- <--- SIP read from UDP:192.168.1.47:5060 --->
- INVITE sip:907611111111@192.168.1.189 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;rport
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189
- CSeq: 20 INVITE
- Call-ID: 0kYkfVYIit
- Max-Forwards: 70
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
- Content-Type: application/sdp
- Content-Length: 465
- Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
- User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
- v=0
- o=6001 3051 3572 IN IP4 192.168.1.47
- s=Talk
- c=IN IP4 192.168.1.47
- t=0 0
- m=audio 7078 RTP/AVP 124 120 111 110 0 8 101
- a=rtpmap:124 opus/48000
- a=fmtp:124 useinbandfec=1; usedtx=1
- a=rtpmap:120 SILK/16000
- a=rtpmap:111 speex/16000
- a=fmtp:111 vbr=on
- a=rtpmap:110 speex/8000
- a=fmtp:110 vbr=on
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- m=video 9078 RTP/AVP 103 99
- a=rtpmap:103 VP8/90000
- a=rtpmap:99 MP4V-ES/90000
- a=fmtp:99 profile-level-id=3
- <------------->
- --- (13 headers 19 lines) ---
- Sending to 192.168.1.47:5060 (NAT)
- Using INVITE request as basis request - 0kYkfVYIit
- Found peer '6001' for '6001' from 192.168.1.47:5060
- <--- Reliably Transmitting (no NAT) to 192.168.1.47:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;received=192.168.1.47;rport=5060
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189;tag=as78e32904
- Call-ID: 0kYkfVYIit
- CSeq: 20 INVITE
- Server: Asterisk PBX 1.8.32.3
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="70289dde"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '0kYkfVYIit' in 32000 ms (Method: INVITE)
- <--- SIP read from UDP:192.168.1.47:5060 --->
- ACK sip:907611111111@192.168.1.189 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;rport
- Call-ID: 0kYkfVYIit
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: <sip:907611111111@192.168.1.189>;tag=as78e32904
- Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
- Max-Forwards: 70
- CSeq: 20 ACK
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.47:5060 --->
- INVITE sip:907611111111@192.168.1.189 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189
- CSeq: 21 INVITE
- Call-ID: 0kYkfVYIit
- Max-Forwards: 70
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
- Content-Type: application/sdp
- Content-Length: 465
- Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
- User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
- Authorization: Digest realm="asterisk", nonce="70289dde", username="6001", uri="sip:907611111111@192.168.1.189", response="a3491ceee9e186203855e136a41e7f0f"
- v=0
- o=6001 3051 3572 IN IP4 192.168.1.47
- s=Talk
- c=IN IP4 192.168.1.47
- t=0 0
- m=audio 7078 RTP/AVP 124 120 111 110 0 8 101
- a=rtpmap:124 opus/48000
- a=fmtp:124 useinbandfec=1; usedtx=1
- a=rtpmap:120 SILK/16000
- a=rtpmap:111 speex/16000
- a=fmtp:111 vbr=on
- a=rtpmap:110 speex/8000
- a=fmtp:110 vbr=on
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- m=video 9078 RTP/AVP 103 99
- a=rtpmap:103 VP8/90000
- a=rtpmap:99 MP4V-ES/90000
- a=fmtp:99 profile-level-id=3
- <------------->
- --- (14 headers 19 lines) ---
- Sending to 192.168.1.47:5060 (no NAT)
- Using INVITE request as basis request - 0kYkfVYIit
- Found peer '6001' for '6001' from 192.168.1.47:5060
- == Using SIP RTP CoS mark 5
- Found RTP audio format 124
- Found RTP audio format 120
- Found RTP audio format 111
- Found RTP audio format 110
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 101
- Found unknown media description format opus for ID 124
- Found unknown media description format SILK for ID 120
- Found audio description format speex for ID 111
- Found audio description format speex for ID 110
- Found audio description format telephone-event for ID 101
- Found RTP video format 103
- Found RTP video format 99
- Found video description format MP4V-ES for ID 99
- Capabilities: us - 0x4 (ulaw), peer - audio=0x20000020c (ulaw|alaw|speex|speex16)/video=0x500000 (h263p|mpeg4)/text=0x0 (nothing), combined - 0x4 (ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 192.168.1.47:7078
- Looking for 907611111111 in office (domain 192.168.1.189)
- list_route: hop: <sip:6001@192.168.1.47>
- <--- Transmitting (no NAT) to 192.168.1.47:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189
- Call-ID: 0kYkfVYIit
- CSeq: 21 INVITE
- Server: Asterisk PBX 1.8.32.3
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:907611111111@192.168.1.189:5060>
- Content-Length: 0
- ------------>
- -- Executing [907611111111@office:1] Log("SIP/6001-00000008", "NOTICE, Dialing out from "" <6001> to 907611111111 via HT503") in new stack
- [Dec 17 09:01:50] NOTICE[1344]: Ext. 907611111111:1 @ office: Dialing out from "" <6001> to 907611111111 via HT503
- -- Executing [907611111111@office:2] MixMonitor("SIP/6001-00000008", "1450342910-office-907611111111.wav") in new stack
- -- Executing [907611111111@office:3] Set("SIP/6001-00000008", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
- -- Executing [907611111111@office:4] Dial("SIP/6001-00000008", "SIP/ht503fxo/907611111111,60") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 18872
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x800 (g726) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 192.168.1.120:5062:
- INVITE sip:907611111111@192.168.1.120 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- Max-Forwards: 70
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>
- Contact: <sip:6001@192.168.1.189:5060>
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.8.32.3
- Date: Thu, 17 Dec 2015 09:01:50 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 291
- v=0
- o=root 523463639 523463639 IN IP4 192.168.1.189
- s=Asterisk PBX 1.8.32.3
- c=IN IP4 192.168.1.189
- t=0 0
- m=audio 18872 RTP/AVP 0 111 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:111 G726-32/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- -- Called SIP/ht503fxo/907611111111
- == Begin MixMonitor Recording SIP/6001-00000008
- <--- SIP read from UDP:192.168.1.120:5062 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 INVITE
- Supported: replaces, path, timer, eventlist
- User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- <--- SIP read from UDP:192.168.1.47:5060 --->
- CANCEL sip:907611111111@192.168.1.189 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
- Call-ID: 0kYkfVYIit
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189
- Max-Forwards: 70
- CSeq: 21 CANCEL
- User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
- <------------->
- --- (8 headers 0 lines) ---
- Sending to 192.168.1.47:5060 (no NAT)
- <--- Reliably Transmitting (no NAT) to 192.168.1.47:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189;tag=as17389654
- Call-ID: 0kYkfVYIit
- CSeq: 21 INVITE
- Server: Asterisk PBX 1.8.32.3
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- <--- Transmitting (no NAT) to 192.168.1.47:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: sip:907611111111@192.168.1.189;tag=as17389654
- Call-ID: 0kYkfVYIit
- CSeq: 21 CANCEL
- Server: Asterisk PBX 1.8.32.3
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
- Reliably Transmitting (no NAT) to 192.168.1.120:5062:
- CANCEL sip:907611111111@192.168.1.120 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- Max-Forwards: 70
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX 1.8.32.3
- Content-Length: 0
- ---
- Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
- == Spawn extension (office, 907611111111, 4) exited non-zero on 'SIP/6001-00000008'
- == End MixMonitor Recording SIP/6001-00000008
- <--- SIP read from UDP:192.168.1.47:5060 --->
- ACK sip:907611111111@192.168.1.189 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
- Call-ID: 0kYkfVYIit
- From: <sip:6001@192.168.1.189>;tag=RaXJH616h
- To: <sip:907611111111@192.168.1.189>;tag=as17389654
- Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
- Max-Forwards: 70
- CSeq: 21 ACK
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog '0kYkfVYIit' Method: ACK
- <--- SIP read from UDP:192.168.1.120:5062 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>;tag=568004079
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 INVITE
- Supported: replaces, path, timer, eventlist
- User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Transmitting (no NAT) to 192.168.1.120:5062:
- ACK sip:907611111111@192.168.1.120 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- Max-Forwards: 70
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>;tag=568004079
- Contact: <sip:6001@192.168.1.189:5060>
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.8.32.3
- Content-Length: 0
- ---
- Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
- <--- SIP read from UDP:192.168.1.120:5062 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
- From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
- To: <sip:907611111111@192.168.1.120>;tag=568004079
- Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
- CSeq: 102 CANCEL
- Contact: <sip:ht503fxo@192.168.1.120:5062>
- Supported: replaces, path, timer, eventlist
- User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- <--- SIP read from UDP:192.168.1.47:5060 --->
- <------------->
- Really destroying SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' Method: INVITE
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