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Asterisk sip.log

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  1. <--- SIP read from UDP:10.145.45.103:5060 --->
  2. INVITE sip:300@54.197.230.121 SIP/2.0
  3. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  4. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  5. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0
  6. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  7. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  8. To: <sip:300@54.197.230.121>
  9. Contact: "111222"<sip:111222@212.28.74.160:51127;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  10. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  11. CSeq: 32613 INVITE
  12. Content-Type: application/sdp
  13. Content-Length: 718
  14. Max-Forwards: 69
  15. User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
  16. Organization: Doubango Telecom
  17.  
  18. v=0
  19. o=mozilla...THIS_IS_SDPARTA-38.0 235096058417272060 0 IN IP4 54.197.230.121
  20. s=Doubango Telecom - firefox
  21. t=0 0
  22. a=msid-semantic:WMS *
  23. m=audio 30762 RTP/AVP 109 9 0 8
  24. c=IN IP4 54.197.230.121
  25. a=end-of-candidates
  26. a=msid:{96bacd9c-573e-4bba-87ab-e94a08c87513} {235a1d8e-9cf5-44de-9429-11c2f51761a8}
  27. a=rtpmap:109 opus/48000/2
  28. a=rtpmap:9 G722/8000/1
  29. a=rtpmap:0 PCMU/8000
  30. a=rtpmap:8 PCMA/8000
  31. a=ssrc:2626043621 cname:{80c717d5-0b49-439b-a7d2-22c4c9268c13}
  32. a=sendrecv
  33. a=rtcp:30763
  34. a=rtcp-mux
  35. a=ice-ufrag:JguEPubo
  36. a=ice-pwd:dlSnv91Z7Ps3lDZVeOMiZJ1HXv
  37. a=candidate:wXn7vaj2mmwmM6pR 1 UDP 2130706431 54.197.230.121 30762 typ host
  38. a=candidate:wXn7vaj2mmwmM6pR 2 UDP 2130706430 54.197.230.121 30763 typ host
  39. <------------->
  40. --- (15 headers 21 lines) ---
  41. Sending to 10.145.45.103:5060 (NAT)
  42. Sending to 10.145.45.103:5060 (NAT)
  43. Using INVITE request as basis request - 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  44. Found peer '111222' for '111222' from 10.145.45.103:5060
  45. == Using SIP RTP CoS mark 5
  46. Found RTP audio format 109
  47. Found RTP audio format 9
  48. Found RTP audio format 0
  49. Found RTP audio format 8
  50. Found unknown media description format opus for ID 109
  51. Found audio description format G722 for ID 9
  52. Found audio description format PCMU for ID 0
  53. Found audio description format PCMA for ID 8
  54. Capabilities: us - (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  55. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  56. Peer audio RTP is at port 54.197.230.121:30762
  57. Looking for 300 in testcontext (domain 54.197.230.121)
  58. list_route: hop: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  59. list_route: hop: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  60.  
  61. <--- Transmitting (no NAT) to 10.145.45.103:5060 --->
  62. SIP/2.0 100 Trying
  63. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  64. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  65. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  66. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  67. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  68. To: <sip:300@54.197.230.121>
  69. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  70. CSeq: 32613 INVITE
  71. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  72. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  73. Supported: replaces, timer
  74. Contact: <sip:300@10.0.0.87:5060>
  75. Content-Length: 0
  76.  
  77.  
  78. <------------>
  79. -- Executing [300@testcontext:1] Answer("SIP/111222-00000006", "") in new stack
  80. Audio is at 31248
  81. Adding codec 100003 (ulaw) to SDP
  82. Adding codec 100004 (alaw) to SDP
  83.  
  84. <--- Reliably Transmitting (no NAT) to 10.145.45.103:5060 --->
  85. SIP/2.0 200 OK
  86. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  87. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  88. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  89. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  90. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  91. To: <sip:300@54.197.230.121>;tag=as25c24143
  92. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  93. CSeq: 32613 INVITE
  94. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  95. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  96. Supported: replaces, timer
  97. Contact: <sip:300@10.0.0.87:5060>
  98. Content-Type: application/sdp
  99. Content-Length: 211
  100.  
  101. v=0
  102. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  103. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  104. c=IN IP4 10.0.0.87
  105. t=0 0
  106. m=audio 31248 RTP/AVP 0 8
  107. a=rtpmap:0 PCMU/8000
  108. a=rtpmap:8 PCMA/8000
  109. a=ptime:20
  110. a=sendrecv
  111.  
  112. <------------>
  113. Retransmitting #1 (no NAT) to 10.145.45.103:5060:
  114. SIP/2.0 200 OK
  115. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  116. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  117. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  118. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  119. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  120. To: <sip:300@54.197.230.121>;tag=as25c24143
  121. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  122. CSeq: 32613 INVITE
  123. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  125. Supported: replaces, timer
  126. Contact: <sip:300@10.0.0.87:5060>
  127. Content-Type: application/sdp
  128. Content-Length: 211
  129.  
  130. v=0
  131. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  132. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  133. c=IN IP4 10.0.0.87
  134. t=0 0
  135. m=audio 31248 RTP/AVP 0 8
  136. a=rtpmap:0 PCMU/8000
  137. a=rtpmap:8 PCMA/8000
  138. a=ptime:20
  139. a=sendrecv
  140.  
  141. ---
  142. Reliably Transmitting (NAT) to 216.53.4.1:5060:
  143. OPTIONS sip:216.53.4.1 SIP/2.0
  144. Via: SIP/2.0/UDP 10.0.0.87:5060;branch=z9hG4bK5a8b7b6a;rport
  145. Max-Forwards: 70
  146. From: "asterisk" <sip:asterisk@10.0.0.87>;tag=as51c74169
  147. To: <sip:216.53.4.1>
  148. Contact: <sip:asterisk@10.0.0.87:5060>
  149. Call-ID: 5b6392181adb4187478f527429e44a7f@10.0.0.87:5060
  150. CSeq: 102 OPTIONS
  151. User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  152. Date: Wed, 24 Jun 2015 13:09:52 GMT
  153. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  154. Supported: replaces, timer
  155. Content-Length: 0
  156.  
  157.  
  158. ---
  159. Retransmitting #2 (no NAT) to 10.145.45.103:5060:
  160. SIP/2.0 200 OK
  161. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  162. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  163. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  164. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  165. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  166. To: <sip:300@54.197.230.121>;tag=as25c24143
  167. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  168. CSeq: 32613 INVITE
  169. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  170. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  171. Supported: replaces, timer
  172. Contact: <sip:300@10.0.0.87:5060>
  173. Content-Type: application/sdp
  174. Content-Length: 211
  175.  
  176. v=0
  177. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  178. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  179. c=IN IP4 10.0.0.87
  180. t=0 0
  181. m=audio 31248 RTP/AVP 0 8
  182. a=rtpmap:0 PCMU/8000
  183. a=rtpmap:8 PCMA/8000
  184. a=ptime:20
  185. a=sendrecv
  186.  
  187. ---
  188. -- Executing [300@testcontext:2] MusicOnHold("SIP/111222-00000006", "") in new stack
  189. -- Started music on hold, class 'default', on SIP/111222-00000006
  190. Retransmitting #3 (no NAT) to 10.145.45.103:5060:
  191. SIP/2.0 200 OK
  192. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  193. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  194. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  195. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  196. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  197. To: <sip:300@54.197.230.121>;tag=as25c24143
  198. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  199. CSeq: 32613 INVITE
  200. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  201. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  202. Supported: replaces, timer
  203. Contact: <sip:300@10.0.0.87:5060>
  204. Content-Type: application/sdp
  205. Content-Length: 211
  206.  
  207. v=0
  208. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  209. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  210. c=IN IP4 10.0.0.87
  211. t=0 0
  212. m=audio 31248 RTP/AVP 0 8
  213. a=rtpmap:0 PCMU/8000
  214. a=rtpmap:8 PCMA/8000
  215. a=ptime:20
  216. a=sendrecv
  217.  
  218. ---
  219. ---
  220. > 0x7fc8cc004ee0 -- Probation passed - setting RTP source address to 54.197.230.121:30762
  221.  
  222.  
  223. ---
  224. Retransmitting #4 (no NAT) to 10.145.45.103:5060:
  225. SIP/2.0 200 OK
  226. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  227. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  228. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  229. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  230. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  231. To: <sip:300@54.197.230.121>;tag=as25c24143
  232. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  233. CSeq: 32613 INVITE
  234. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  236. Supported: replaces, timer
  237. Contact: <sip:300@10.0.0.87:5060>
  238. Content-Type: application/sdp
  239. Content-Length: 211
  240.  
  241. v=0
  242. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  243. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  244. c=IN IP4 10.0.0.87
  245. t=0 0
  246. m=audio 31248 RTP/AVP 0 8
  247. a=rtpmap:0 PCMU/8000
  248. a=rtpmap:8 PCMA/8000
  249. a=ptime:20
  250. a=sendrecv
  251.  
  252. ---
  253.  
  254.  
  255. ---
  256. Retransmitting #5 (no NAT) to 10.145.45.103:5060:
  257. SIP/2.0 200 OK
  258. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  259. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  260. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  261. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  262. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  263. To: <sip:300@54.197.230.121>;tag=as25c24143
  264. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  265. CSeq: 32613 INVITE
  266. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  268. Supported: replaces, timer
  269. Contact: <sip:300@10.0.0.87:5060>
  270. Content-Type: application/sdp
  271. Content-Length: 211
  272.  
  273. v=0
  274. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  275. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  276. c=IN IP4 10.0.0.87
  277. t=0 0
  278. m=audio 31248 RTP/AVP 0 8
  279. a=rtpmap:0 PCMU/8000
  280. a=rtpmap:8 PCMA/8000
  281. a=ptime:20
  282. a=sendrecv
  283.  
  284. ---
  285. Really destroying SIP dialog '286ae7d7689a718a6fd2175e08960ef3@10.0.0.87:5060' Method: OPTIONS
  286. Retransmitting #6 (no NAT) to 10.145.45.103:5060:
  287. SIP/2.0 200 OK
  288. Via: SIP/2.0/UDP 54.197.230.121:5060;branch=z9hG4bK1778.dae9d5a083cae2d05d8587f57941f02c.0;received=10.145.45.103
  289. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=212.28.74.160;branch=z9hG4bK6JLG4LGIBASafyfuw9glTMFh6pqlFmXA;rport=51127
  290. Record-Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  291. Record-Route: <sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  292. From: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  293. To: <sip:300@54.197.230.121>;tag=as25c24143
  294. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  295. CSeq: 32613 INVITE
  296. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  297. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  298. Supported: replaces, timer
  299. Contact: <sip:300@10.0.0.87:5060>
  300. Content-Type: application/sdp
  301. Content-Length: 211
  302.  
  303. v=0
  304. o=root 2101314398 2101314398 IN IP4 10.0.0.87
  305. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  306. c=IN IP4 10.0.0.87
  307. t=0 0
  308. m=audio 31248 RTP/AVP 0 8
  309. a=rtpmap:0 PCMU/8000
  310. a=rtpmap:8 PCMA/8000
  311. a=ptime:20
  312. a=sendrecv
  313.  
  314. ---
  315. [2015-06-24 13:09:58] WARNING[28692]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48 for seqno 32613 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  316. Packet timed out after 6400ms with no response
  317. [2015-06-24 13:09:58] WARNING[28692]: chan_sip.c:4204 retrans_pkt: Hanging up call 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  318. -- Stopped music on hold on SIP/111222-00000006
  319. == Spawn extension (testcontext, 300, 2) exited non-zero on 'SIP/111222-00000006'
  320. Scheduling destruction of SIP dialog '8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48' in 6400 ms (Method: INVITE)
  321. set_destination: Parsing <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes> for address/port to send to
  322. set_destination: set destination to 54.197.230.121:5060
  323. Reliably Transmitting (no NAT) to 54.197.230.121:5060:
  324. BYE sip:111222@212.28.74.160:51127;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
  325. Via: SIP/2.0/UDP 10.0.0.87:5060;branch=z9hG4bK323f8d72
  326. Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>,<sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  327. Max-Forwards: 70
  328. From: <sip:300@54.197.230.121>;tag=as25c24143
  329. To: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  330. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  331. CSeq: 102 BYE
  332. User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  333. X-Asterisk-HangupCause: No user responding
  334. X-Asterisk-HangupCauseCode: 18
  335. Content-Length: 0
  336.  
  337.  
  338. ---
  339. Retransmitting #1 (no NAT) to 54.197.230.121:5060:
  340. BYE sip:111222@212.28.74.160:51127;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
  341. Via: SIP/2.0/UDP 10.0.0.87:5060;branch=z9hG4bK323f8d72
  342. Route: <sip:54.197.230.121:5060;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>,<sip:54.197.230.121:10080;transport=ws;r2=on;lr=on;ftag=CuBgeTFstNchchFIHEOs;nat=yes>
  343. Max-Forwards: 70
  344. From: <sip:300@54.197.230.121>;tag=as25c24143
  345. To: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  346. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  347. CSeq: 102 BYE
  348. User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  349. X-Asterisk-HangupCause: No user responding
  350. X-Asterisk-HangupCauseCode: 18
  351. Content-Length: 0
  352.  
  353.  
  354. ---
  355.  
  356. <--- SIP read from UDP:54.197.230.121:5060 --->
  357. SIP/2.0 200 OK
  358. From: <sip:300@54.197.230.121>;tag=as25c24143
  359. To: "111222"<sip:111222@54.197.230.121>;tag=CuBgeTFstNchchFIHEOs
  360. Contact: <sip:111222@df7jal23ls0d.invalid;alias=212.28.74.160~51127~5;transport=ws>
  361. Call-ID: 8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48
  362. CSeq: 102 BYE
  363. Content-Length: 0
  364. Via: SIP/2.0/UDP 10.0.0.87:5060;rport=5060;received=54.164.207.254;branch=z9hG4bK323f8d72
  365.  
  366. <------------->
  367. --- (8 headers 0 lines) ---
  368. SIP Response message for INCOMING dialog BYE arrived
  369. Really destroying SIP dialog '8ed8365e-3ed9-52b8-bf1d-b13a7f61cc48' Method: INVITE
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