ilumen

redirect

Jan 2nd, 2018
425
0
Never
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
  1. LI>
  2.  
  3. <--- SIP read from UDP:195.128.182.62:5060 --->
  4. INVITE sip:442288505@172.31.1.100:5060 SIP/2.0
  5. Call-ID: 135aec32753b47e340e04ae2a947e8
  6. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b
  7. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  8. To: "442288505" <sip:442288505@sip.intertelecom.ua>
  9. CSeq: 25021 INVITE
  10. Max-Forwards: 70
  11. User-Agent: Smile CTI Server
  12. X-Redirecting-Number: 442288505
  13. Contact: <sip:[email protected]>
  14. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, REFER, NOTIFY, INFO, SUBSCRIBE, MESSAGE
  15. Content-Type: application/sdp
  16. Content-Length: 330
  17.  
  18. v=0
  19. o=0443601932 1514907978 1 IN IP4 195.128.182.62
  20. s=SIP Call
  21. c=IN IP4 195.128.182.62
  22. t=0 0
  23. m=audio 63048 RTP/AVP 8 0 18 4 110 101
  24. a=rtpmap:8 PCMA/8000
  25. a=rtpmap:0 PCMU/8000
  26. a=rtpmap:18 G729/8000
  27. a=rtpmap:4 G723/8000
  28. a=rtpmap:110 opus/48000/2
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-15
  31. a=ptime:20
  32. a=sendrecv
  33. <------------->
  34. --- (13 headers 15 lines) ---
  35. Sending to 195.128.182.62:5060 (no NAT)
  36. Sending to 195.128.182.62:5060 (no NAT)
  37. Using INVITE request as basis request - 135aec32753b47e340e04ae2a947e8
  38. Found peer 'intertelecom-prop' for '0443601932' from 195.128.182.62:5060
  39.   == Using SIP RTP CoS mark 5
  40. Found RTP audio format 8
  41. Found RTP audio format 0
  42. Found RTP audio format 18
  43. Found RTP audio format 4
  44. Found RTP audio format 110
  45. Found RTP audio format 101
  46. Found audio description format PCMA for ID 8
  47. Found audio description format PCMU for ID 0
  48. Found audio description format G729 for ID 18
  49. Found audio description format G723 for ID 4
  50. Found audio description format opus for ID 110
  51. Found audio description format telephone-event for ID 101
  52. Capabilities: us - (ulaw|alaw|gsm|h263|g729), peer - audio=(ulaw|g723|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
  53. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  54. Peer audio RTP is at port 195.128.182.62:63048
  55. Looking for 442288505 in intertelecom-incoming-prop (domain 172.31.1.100)
  56. sip_route_dump: route/path hop: <sip:[email protected]>
  57.  
  58. <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
  59. SIP/2.0 100 Trying
  60. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
  61. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  62. To: "442288505" <sip:442288505@sip.intertelecom.ua>
  63. Call-ID: 135aec32753b47e340e04ae2a947e8
  64. CSeq: 25021 INVITE
  65. Server: Asterisk PBX 13.18.5
  66. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  67. Supported: replaces, timer
  68. Contact: <sip:442288505@172.31.1.100:5060>
  69. Content-Length: 0
  70.  
  71.  
  72. <------------>
  73.     -- Executing [442288505@intertelecom-incoming-prop:1] Macro("SIP/intertelecom-prop-0000000a", "recording,0443601932,442288505") in new stack
  74.     -- Executing [s@macro-recording:1] GotoIf("SIP/intertelecom-prop-0000000a", "1?mp3:no") in new stack
  75.     -- Goto (macro-recording,s,3)
  76.     -- Executing [s@macro-recording:3] Set("SIP/intertelecom-prop-0000000a", "fname=1514907978.27-2018-01-02-17_46-0443601932-442288505") in new stack
  77.     -- 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
  78.     -- 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
  79.     -- Executing [s@macro-recording:6] Set("SIP/intertelecom-prop-0000000a", "CDR(realdst)=442288505") in new stack
  80.     -- Executing [s@macro-recording:7] Set("SIP/intertelecom-prop-0000000a", "CDR(remoteip)=195.128.182.62") in new stack
  81.     -- 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
  82.     -- Executing [s@macro-recording:9] Goto("SIP/intertelecom-prop-0000000a", "no") in new stack
  83.     -- Goto (macro-recording,s,16)
  84.     -- Executing [s@macro-recording:16] Verbose("SIP/intertelecom-prop-0000000a", "Exit record") in new stack
  85.   == Begin MixMonitor Recording SIP/intertelecom-prop-0000000a
  86. Exit record
  87.     -- Executing [442288505@intertelecom-incoming-prop:2] Dial("SIP/intertelecom-prop-0000000a", "SIP/204,600,rRt") in new stack
  88.   == Using SIP RTP CoS mark 5
  89. Audio is at 14958
  90. Adding codec ulaw to SDP
  91. Adding codec alaw to SDP
  92. Adding non-codec 0x1 (telephone-event) to SDP
  93. Reliably Transmitting (NAT) to 91.196.100.143:1035:
  94. INVITE sip:204@192.168.40.22:5060 SIP/2.0
  95. Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport
  96. Max-Forwards: 70
  97. From: "0443601932" <sip:[email protected]>;tag=as61581287
  98. To: <sip:204@192.168.40.22:5060>
  99. Contact: <sip:[email protected]:5060>
  100. Call-ID: [email protected]:5060
  101. CSeq: 102 INVITE
  102. User-Agent: Asterisk PBX 13.18.5
  103. Date: Tue, 02 Jan 2018 15:46:18 GMT
  104. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  105. Supported: replaces, timer
  106. Content-Type: application/sdp
  107. Content-Length: 262
  108.  
  109. v=0
  110. o=root 664565165 664565165 IN IP4 172.31.1.100
  111. s=Asterisk PBX 13.18.5
  112. c=IN IP4 172.31.1.100
  113. t=0 0
  114. m=audio 14958 RTP/AVP 0 8 101
  115. a=rtpmap:0 PCMU/8000
  116. a=rtpmap:8 PCMA/8000
  117. a=rtpmap:101 telephone-event/8000
  118. a=fmtp:101 0-16
  119. a=maxptime:150
  120. a=sendrecv
  121.  
  122. ---
  123.     -- Called SIP/204
  124.  
  125. <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
  126. SIP/2.0 180 Ringing
  127. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
  128. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  129. To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
  130. Call-ID: 135aec32753b47e340e04ae2a947e8
  131. CSeq: 25021 INVITE
  132. Server: Asterisk PBX 13.18.5
  133. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  134. Supported: replaces, timer
  135. Contact: <sip:442288505@172.31.1.100:5060>
  136. Content-Length: 0
  137.  
  138.  
  139. <------------>
  140.  
  141. <--- SIP read from UDP:91.196.100.143:1035 --->
  142. SIP/2.0 100 Trying
  143. Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport=5060
  144. From: "0443601932" <sip:[email protected]:5060>;tag=as61581287
  145. To: <sip:204@192.168.40.22:5060>
  146. Call-ID: [email protected]:5060
  147. CSeq: 102 INVITE
  148. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  149. Content-Length: 0
  150.  
  151. <------------->
  152. --- (8 headers 0 lines) ---
  153.  
  154. <--- SIP read from UDP:91.196.100.143:1035 --->
  155. SIP/2.0 302 Moved Temporarily
  156. Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport=5060
  157. From: "0443601932" <sip:[email protected]:5060>;tag=as61581287
  158. To: <sip:204@192.168.40.22:5060>;tag=2600424852
  159. Call-ID: [email protected]:5060
  160. CSeq: 102 INVITE
  161. Contact: "0950795899" <sip:[email protected]:5060>
  162. User-Agent: DLINK DPH-150SE FRU2.2.1328.545
  163. Diversion: <sip:204@192.168.40.22:5060>;reason=unconditional
  164. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  165. Content-Length: 0
  166.  
  167. <------------->
  168. --- (11 headers 0 lines) ---
  169.     -- Got SIP response 302 "Moved Temporarily" back from 91.196.100.143:1035
  170. RDNIS for this call is 204 (reason unconditional)
  171. Transmitting (NAT) to 91.196.100.143:1035:
  172. ACK sip:204@192.168.40.22:5060 SIP/2.0
  173. Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK47cf517f;rport
  174. Max-Forwards: 70
  175. From: "0443601932" <sip:[email protected]>;tag=as61581287
  176. To: <sip:204@192.168.40.22:5060>;tag=2600424852
  177. Contact: <sip:[email protected]:5060>
  178. Call-ID: [email protected]:5060
  179. CSeq: 102 ACK
  180. User-Agent: Asterisk PBX 13.18.5
  181. Content-Length: 0
  182.  
  183.  
  184. ---
  185.     -- Now forwarding SIP/intertelecom-prop-0000000a to 'Local/0950795899@prop' (thanks to SIP/204-0000000b)
  186. [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
  187.  
  188. <--- Transmitting (no NAT) to 195.128.182.62:5060 --->
  189. SIP/2.0 181 Call is being forwarded
  190. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
  191. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  192. To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
  193. Call-ID: 135aec32753b47e340e04ae2a947e8
  194. CSeq: 25021 INVITE
  195. Server: Asterisk PBX 13.18.5
  196. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  197. Supported: replaces, timer
  198. Contact: <sip:442288505@172.31.1.100:5060>
  199. Diversion: <sip:204@172.31.1.100>;reason=unconditional
  200. Content-Length: 0
  201.  
  202.  
  203. <------------>
  204.     -- Executing [0950795899@prop:1] Macro("Local/0950795899@prop-00000004;2", "recording,0443601932,0950795899") in new stack
  205.     -- Executing [s@macro-recording:1] GotoIf("Local/0950795899@prop-00000004;2", "1?mp3:no") in new stack
  206.     -- Goto (macro-recording,s,3)
  207.     -- Executing [s@macro-recording:3] Set("Local/0950795899@prop-00000004;2", "fname=1514907978.30-2018-01-02-17_46-0443601932-0950795899") in new stack
  208.     -- 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
  209.     -- 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
  210.     -- Executing [s@macro-recording:6] Set("Local/0950795899@prop-00000004;2", "CDR(realdst)=0950795899") in new stack
  211. [Jan  2 17:46:18] WARNING[18675][C-00000005]: func_channel.c:465 func_channel_read: Unknown or unavailable item requested: 'recvip'
  212.     -- Executing [s@macro-recording:7] Set("Local/0950795899@prop-00000004;2", "CDR(remoteip)=") in new stack
  213.     -- 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
  214.     -- Executing [s@macro-recording:9] Goto("Local/0950795899@prop-00000004;2", "no") in new stack
  215.     -- Goto (macro-recording,s,16)
  216.     -- Executing [s@macro-recording:16] Verbose("Local/0950795899@prop-00000004;2", "Exit record") in new stack
  217. Exit record
  218.     -- Auto fallthrough, channel 'Local/0950795899@prop-00000004;2' status is 'UNKNOWN'
  219.   == Begin MixMonitor Recording Local/0950795899@prop-00000004;2
  220.   == Everyone is busy/congested at this time (1:0/0/1)
  221.   == MixMonitor close filestream (mixed)
  222.   == 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"]
  223.     -- Executing [442288505@intertelecom-incoming-prop:3] Hangup("SIP/intertelecom-prop-0000000a", "") in new stack
  224.   == Spawn extension (intertelecom-incoming-prop, 442288505, 3) exited non-zero on 'SIP/intertelecom-prop-0000000a'
  225. Scheduling destruction of SIP dialog '135aec32753b47e340e04ae2a947e8' in 32000 ms (Method: INVITE)
  226.  
  227. <--- Reliably Transmitting (no NAT) to 195.128.182.62:5060 --->
  228. SIP/2.0 603 Declined
  229. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b;received=195.128.182.62
  230. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  231. To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
  232. Call-ID: 135aec32753b47e340e04ae2a947e8
  233. CSeq: 25021 INVITE
  234. Server: Asterisk PBX 13.18.5
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  236. Supported: replaces, timer
  237. Content-Length: 0
  238.  
  239.  
  240. <------------>
  241. Really destroying SIP dialog '[email protected]:5060' Method: INVITE
  242.   == MixMonitor close filestream (mixed)
  243.   == 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"]
  244.   == End MixMonitor Recording Local/0950795899@prop-00000004;2
  245.   == End MixMonitor Recording SIP/intertelecom-prop-0000000a
  246.  
  247. <--- SIP read from UDP:195.128.182.62:5060 --->
  248. ACK sip:442288505@172.31.1.100:5060 SIP/2.0
  249. Call-ID: 135aec32753b47e340e04ae2a947e8
  250. Via: SIP/2.0/UDP sip.intertelecom.ua:5060;branch=z9hG4bKb4aef5b0f746924b
  251. From: "0443601932" <sip:[email protected]>;tag=c9584b06
  252. To: "442288505" <sip:442288505@sip.intertelecom.ua>;tag=as7fc1d10a
  253. CSeq: 25021 ACK
  254. Max-Forwards: 70
  255. User-Agent: Smile CTI Server
  256. X-Redirecting-Number: 442288505
  257. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, REFER, NOTIFY, INFO, SUBSCRIBE, MESSAGE
  258. Content-Length: 0
  259.  
  260. <------------->
  261. --- (11 headers 0 lines) ---
  262. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  263.  
  264. <--- SIP read from UDP:91.196.100.143:1033 --->
  265. OPTIONS sip:172.31.1.100:5060 SIP/2.0
  266. Via: SIP/2.0/UDP 192.168.40.12:5060;branch=z9hG4bK152729778530422316
  267. From: 401 <sip:401@172.31.1.100:5060>;tag=81607423
  268. To: <sip:172.31.1.100:5060>
  269. Call-ID: 23968165421877-249662124924419@192.168.40.12
  270. CSeq: 1 OPTIONS
  271. Max-Forwards: 70
  272. User-Agent: DLINK DPH-150S FRU2.2.1556.591
  273. Accept: application/sdp
  274. Content-Length: 0
  275.  
  276. <------------->
  277. --- (10 headers 0 lines) ---
  278. Sending to 91.196.100.143:1033 (NAT)
  279. Looking for s in default (domain 172.31.1.100)
  280.  
  281. <--- Transmitting (NAT) to 91.196.100.143:1033 --->
  282. SIP/2.0 404 Not Found
  283. Via: SIP/2.0/UDP 192.168.40.12:5060;branch=z9hG4bK152729778530422316;received=91.196.100.143;rport=1033
  284. From: 401 <sip:401@172.31.1.100:5060>;tag=81607423
  285. To: <sip:172.31.1.100:5060>;tag=as02097a6a
  286. Call-ID: 23968165421877-249662124924419@192.168.40.12
  287. CSeq: 1 OPTIONS
  288. Server: Asterisk PBX 13.18.5
  289. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  290. Supported: replaces, timer
  291. Accept: application/sdp
  292. Content-Length: 0
  293.  
  294.  
  295. <------------>
  296. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
Advertisement
Add Comment
Please, Sign In to add comment