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- <--- SIP read from UDP:195.128.182.62:5060 --->
- INVITE sip:442288505@172.31.1.100:5060 SIP/2.0
- Call-ID: 135aec32753b47e340e04ae2a947e8
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>
- CSeq: 25021 INVITE
- Max-Forwards: 70
- User-Agent: Smile CTI Server
- X-Redirecting-Number: 442288505
- Contact: <sip:[email protected]>
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, REFER, NOTIFY, INFO, SUBSCRIBE, MESSAGE
- Content-Type: application/sdp
- Content-Length: 330
- v=0
- o=0443601932 1514907978 1 IN IP4 195.128.182.62
- s=SIP Call
- c=IN IP4 195.128.182.62
- t=0 0
- m=audio 63048 RTP/AVP 8 0 18 4 110 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=rtpmap:4 G723/8000
- a=rtpmap:110 opus/48000/2
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:20
- a=sendrecv
- <------------->
- --- (13 headers 15 lines) ---
- Sending to 195.128.182.62:5060 (no NAT)
- Sending to 195.128.182.62:5060 (no NAT)
- Using INVITE request as basis request - 135aec32753b47e340e04ae2a947e8
- Found peer 'intertelecom-prop' for '0443601932' from 195.128.182.62:5060
- == Using SIP RTP CoS mark 5
- Found RTP audio format 8
- Found RTP audio format 0
- Found RTP audio format 18
- Found RTP audio format 4
- Found RTP audio format 110
- 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 G729 for ID 18
- Found audio description format G723 for ID 4
- Found audio description format opus for ID 110
- Found audio description format telephone-event for ID 101
- Capabilities: us - (ulaw|alaw|gsm|h263|g729), peer - audio=(ulaw|g723|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 195.128.182.62:63048
- Looking for 442288505 in intertelecom-incoming-prop (domain 172.31.1.100)
- sip_route_dump: route/path hop: <sip:[email protected]>
- <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>
- Call-ID: 135aec32753b47e340e04ae2a947e8
- CSeq: 25021 INVITE
- Server: Asterisk PBX 13.18.5
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:442288505@172.31.1.100:5060>
- Content-Length: 0
- <------------>
- -- Executing [442288505@intertelecom-incoming-prop:1] Macro("SIP/intertelecom-prop-0000000a", "recording,0443601932,442288505") in new stack
- -- Executing [s@macro-recording:1] GotoIf("SIP/intertelecom-prop-0000000a", "1?mp3:no") in new stack
- -- Goto (macro-recording,s,3)
- -- Executing [s@macro-recording:3] Set("SIP/intertelecom-prop-0000000a", "fname=1514907978.27-2018-01-02-17_46-0443601932-442288505") in new stack
- -- Executing [s@macro-recording:4] Set("SIP/intertelecom-prop-0000000a", "monopt=nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3"") in new stack
- -- Executing [s@macro-recording:5] Set("SIP/intertelecom-prop-0000000a", "CDR(filename)=1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3") in new stack
- -- Executing [s@macro-recording:6] Set("SIP/intertelecom-prop-0000000a", "CDR(realdst)=442288505") in new stack
- -- Executing [s@macro-recording:7] Set("SIP/intertelecom-prop-0000000a", "CDR(remoteip)=195.128.182.62") in new stack
- -- Executing [s@macro-recording:8] MixMonitor("SIP/intertelecom-prop-0000000a", "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav,b,nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3"") in new stack
- -- Executing [s@macro-recording:9] Goto("SIP/intertelecom-prop-0000000a", "no") in new stack
- -- Goto (macro-recording,s,16)
- -- Executing [s@macro-recording:16] Verbose("SIP/intertelecom-prop-0000000a", "Exit record") in new stack
- == Begin MixMonitor Recording SIP/intertelecom-prop-0000000a
- Exit record
- -- Executing [442288505@intertelecom-incoming-prop:2] Dial("SIP/intertelecom-prop-0000000a", "SIP/204,600,rRt") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 14958
- Adding codec ulaw to SDP
- Adding codec alaw to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 91.196.100.143:1035:
- INVITE sip:204@192.168.40.22:5060 SIP/2.0
- Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport
- Max-Forwards: 70
- From: "0443601932" <sip:[email protected]>;tag=as61581287
- To: <sip:204@192.168.40.22:5060>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 13.18.5
- Date: Tue, 02 Jan 2018 15:46:18 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 262
- v=0
- o=root 664565165 664565165 IN IP4 172.31.1.100
- s=Asterisk PBX 13.18.5
- c=IN IP4 172.31.1.100
- t=0 0
- m=audio 14958 RTP/AVP 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=maxptime:150
- a=sendrecv
- ---
- -- Called SIP/204
- <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
- Call-ID: 135aec32753b47e340e04ae2a947e8
- CSeq: 25021 INVITE
- Server: Asterisk PBX 13.18.5
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:442288505@172.31.1.100:5060>
- Content-Length: 0
- <------------>
- <--- SIP read from UDP:91.196.100.143:1035 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport=5060
- To: <sip:204@192.168.40.22:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:91.196.100.143:1035 --->
- SIP/2.0 302 Moved Temporarily
- Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport=5060
- To: <sip:204@192.168.40.22:5060>;tag=2600424852
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- User-Agent: DLINK DPH-150SE FRU2.2.1328.545
- Diversion: <sip:204@192.168.40.22:5060>;reason=unconditional
- Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- -- Got SIP response 302 "Moved Temporarily" back from 91.196.100.143:1035
- RDNIS for this call is 204 (reason unconditional)
- Transmitting (NAT) to 91.196.100.143:1035:
- ACK sip:204@192.168.40.22:5060 SIP/2.0
- Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport
- Max-Forwards: 70
- From: "0443601932" <sip:[email protected]>;tag=as61581287
- To: <sip:204@192.168.40.22:5060>;tag=2600424852
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 13.18.5
- Content-Length: 0
- ---
- -- Now forwarding SIP/intertelecom-prop-0000000a to 'Local/0950795899@prop' (thanks to SIP/204-0000000b)
- [Jan 2 17:46:18] NOTICE[18673][C-00000005]: app_dial.c:1000 do_forward: Not accepting call completion offers from call-forward recipient Local/0950795899@prop-00000004;1
- <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
- SIP/2.0 181 Call is being forwarded
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
- Call-ID: 135aec32753b47e340e04ae2a947e8
- CSeq: 25021 INVITE
- Server: Asterisk PBX 13.18.5
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:442288505@172.31.1.100:5060>
- Diversion: <sip:204@172.31.1.100>;reason=unconditional
- Content-Length: 0
- <------------>
- -- Executing [0950795899@prop:1] Macro("Local/0950795899@prop-00000004;2", "recording,0443601932,0950795899") in new stack
- -- Executing [s@macro-recording:1] GotoIf("Local/0950795899@prop-00000004;2", "1?mp3:no") in new stack
- -- Goto (macro-recording,s,3)
- -- Executing [s@macro-recording:3] Set("Local/0950795899@prop-00000004;2", "fname=1514907978.30-2018-01-02-17_46-0443601932-0950795899") in new stack
- -- Executing [s@macro-recording:4] Set("Local/0950795899@prop-00000004;2", "monopt=nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3"") in new stack
- -- Executing [s@macro-recording:5] Set("Local/0950795899@prop-00000004;2", "CDR(filename)=1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3") in new stack
- -- Executing [s@macro-recording:6] Set("Local/0950795899@prop-00000004;2", "CDR(realdst)=0950795899") in new stack
- [Jan 2 17:46:18] WARNING[18675][C-00000005]: func_channel.c:465 func_channel_read: Unknown or unavailable item requested: 'recvip'
- -- Executing [s@macro-recording:7] Set("Local/0950795899@prop-00000004;2", "CDR(remoteip)=") in new stack
- -- Executing [s@macro-recording:8] MixMonitor("Local/0950795899@prop-00000004;2", "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav,b,nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3"") in new stack
- -- Executing [s@macro-recording:9] Goto("Local/0950795899@prop-00000004;2", "no") in new stack
- -- Goto (macro-recording,s,16)
- -- Executing [s@macro-recording:16] Verbose("Local/0950795899@prop-00000004;2", "Exit record") in new stack
- Exit record
- -- Auto fallthrough, channel 'Local/0950795899@prop-00000004;2' status is 'UNKNOWN'
- == Begin MixMonitor Recording Local/0950795899@prop-00000004;2
- == Everyone is busy/congested at this time (1:0/0/1)
- == MixMonitor close filestream (mixed)
- == Executing [nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.30-2018-01-02-17_46-0443601932-0950795899.mp3"]
- -- Executing [442288505@intertelecom-incoming-prop:3] Hangup("SIP/intertelecom-prop-0000000a", "") in new stack
- == Spawn extension (intertelecom-incoming-prop, 442288505, 3) exited non-zero on 'SIP/intertelecom-prop-0000000a'
- Scheduling destruction of SIP dialog '135aec32753b47e340e04ae2a947e8' in 32000 ms (Method: INVITE)
- <--- Reliably Transmitting (no NAT) to 195.128.182.62:5060 --->
- SIP/2.0 603 Declined
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
- Call-ID: 135aec32753b47e340e04ae2a947e8
- CSeq: 25021 INVITE
- Server: Asterisk PBX 13.18.5
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Really destroying SIP dialog '[email protected]:5060' Method: INVITE
- == MixMonitor close filestream (mixed)
- == Executing [nice -n 19 /usr/local/bin/lame -b 32 --silent "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3" && rm -f "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.wav" && chmod o+r "/var/spool/asterisk/monitor/1514907978.27-2018-01-02-17_46-0443601932-442288505.mp3"]
- == End MixMonitor Recording Local/0950795899@prop-00000004;2
- == End MixMonitor Recording SIP/intertelecom-prop-0000000a
- <--- SIP read from UDP:195.128.182.62:5060 --->
- ACK sip:442288505@172.31.1.100:5060 SIP/2.0
- Call-ID: 135aec32753b47e340e04ae2a947e8
- Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b
- From: "0443601932" <sip:[email protected]>;tag=c9584b06
- To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
- CSeq: 25021 ACK
- Max-Forwards: 70
- User-Agent: Smile CTI Server
- X-Redirecting-Number: 442288505
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, REFER, NOTIFY, INFO, SUBSCRIBE, MESSAGE
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:91.196.100.143:1033 --->
- OPTIONS sip:172.31.1.100:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.40.12:5060;branch=z9hG4bK152729778530422316
- From: 401 <sip:401@172.31.1.100:5060>;tag=81607423
- To: <sip:172.31.1.100:5060>
- Call-ID: 23968165421877-249662124924419@192.168.40.12
- CSeq: 1 OPTIONS
- Max-Forwards: 70
- User-Agent: DLINK DPH-150S FRU2.2.1556.591
- Accept: application/sdp
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Sending to 91.196.100.143:1033 (NAT)
- Looking for s in default (domain 172.31.1.100)
- <--- Transmitting (NAT) to 91.196.100.143:1033 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 192.168.40.12:5060;branch=z9hG4bK152729778530422316;received=91.196.100.143;rport=1033
- From: 401 <sip:401@172.31.1.100:5060>;tag=81607423
- To: <sip:172.31.1.100:5060>;tag=as02097a6a
- Call-ID: 23968165421877-249662124924419@192.168.40.12
- CSeq: 1 OPTIONS
- Server: Asterisk PBX 13.18.5
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Accept: application/sdp
- Content-Length: 0
- <------------>
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