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  1. <--- SIP read from UDP:97.122.234.169:5061 --->
  2. INVITE sip:100@40.122.50.92;transport=UDP SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;rport
  4. Max-Forwards: 70
  5. Contact: <sip:6001@192.168.0.94:5061;transport=UDP>
  6. To: <sip:100@40.122.50.92;transport=UDP>
  7. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  8. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  9. CSeq: 1 INVITE
  10. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  11. Content-Type: application/sdp
  12. User-Agent: Z 5.2.28 rv2.8.114
  13. Allow-Events: presence, kpml, talk
  14. Content-Length: 608
  15.  
  16. v=0
  17. o=Z 401744695 0 IN IP4 192.168.0.94
  18. s=Z
  19. c=IN IP4 192.168.0.94
  20. t=0 0
  21. m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
  22. a=rtpmap:106 opus/48000/2
  23. a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
  24. a=rtpmap:111 speex/16000
  25. a=rtpmap:97 iLBC/8000
  26. a=fmtp:97 mode=20
  27. a=rtpmap:110 speex/8000
  28. a=rtpmap:112 speex/32000
  29. a=rtpmap:98 telephone-event/48000
  30. a=fmtp:98 0-16
  31. a=rtpmap:101 telephone-event/8000
  32. a=fmtp:101 0-16
  33. a=rtpmap:100 telephone-event/16000
  34. a=fmtp:100 0-16
  35. a=rtpmap:99 telephone-event/32000
  36. a=fmtp:99 0-16
  37. a=rtpmap:102 G726-32/8000
  38. a=sendrecv
  39. <------------->
  40. --- (13 headers 23 lines) ---
  41. Sending to 97.122.234.169:5061 (NAT)
  42. Sending to 97.122.234.169:5061 (NAT)
  43. Using INVITE request as basis request - kv40PFslKyjCAY7ZiEL9kA..
  44. Found peer '6001' for '6001' from 97.122.234.169:5061
  45.  
  46. <--- Reliably Transmitting (NAT) to 97.122.234.169:5061 --->
  47. SIP/2.0 401 Unauthorized
  48. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;received=97.122.234.169;rport=5061
  49. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  50. To: <sip:100@40.122.50.92;transport=UDP>;tag=as191bd6c4
  51. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  52. CSeq: 1 INVITE
  53. Server: Asterisk PBX 16.5.0
  54. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  55. Supported: replaces, timer
  56. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02597d71"
  57. Content-Length: 0
  58.  
  59.  
  60. <------------>
  61. Scheduling destruction of SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' in 32000 ms (Method: INVITE)
  62.  
  63. <--- SIP read from UDP:97.122.234.169:5061 --->
  64. ACK sip:100@40.122.50.92;transport=UDP SIP/2.0
  65. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;rport
  66. Max-Forwards: 70
  67. To: <sip:100@40.122.50.92;transport=UDP>;tag=as191bd6c4
  68. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  69. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  70. CSeq: 1 ACK
  71. Content-Length: 0
  72.  
  73. <------------->
  74. --- (8 headers 0 lines) ---
  75.  
  76. <--- SIP read from UDP:97.122.234.169:5061 --->
  77. INVITE sip:100@40.122.50.92;transport=UDP SIP/2.0
  78. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;rport
  79. Max-Forwards: 70
  80. Contact: <sip:6001@192.168.0.94:5061;transport=UDP>
  81. To: <sip:100@40.122.50.92;transport=UDP>
  82. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  83. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  84. CSeq: 2 INVITE
  85. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  86. Content-Type: application/sdp
  87. User-Agent: Z 5.2.28 rv2.8.114
  88. Authorization: Digest username="6001",realm="asterisk",nonce="02597d71",uri="sip:[email protected];transport=UDP",response="1858388598874fd6c70b5015c49289f0",algorithm=MD5
  89. Allow-Events: presence, kpml, talk
  90. Content-Length: 608
  91.  
  92. v=0
  93. o=Z 401744695 0 IN IP4 192.168.0.94
  94. s=Z
  95. c=IN IP4 192.168.0.94
  96. t=0 0
  97. m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
  98. a=rtpmap:106 opus/48000/2
  99. a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
  100. a=rtpmap:111 speex/16000
  101. a=rtpmap:97 iLBC/8000
  102. a=fmtp:97 mode=20
  103. a=rtpmap:110 speex/8000
  104. a=rtpmap:112 speex/32000
  105. a=rtpmap:98 telephone-event/48000
  106. a=fmtp:98 0-16
  107. a=rtpmap:101 telephone-event/8000
  108. a=fmtp:101 0-16
  109. a=rtpmap:100 telephone-event/16000
  110. a=fmtp:100 0-16
  111. a=rtpmap:99 telephone-event/32000
  112. a=fmtp:99 0-16
  113. a=rtpmap:102 G726-32/8000
  114. a=sendrecv
  115. <------------->
  116. --- (14 headers 23 lines) ---
  117. Sending to 97.122.234.169:5061 (NAT)
  118. Using INVITE request as basis request - kv40PFslKyjCAY7ZiEL9kA..
  119. Found peer '6001' for '6001' from 97.122.234.169:5061
  120.   == Using SIP RTP CoS mark 5
  121. Found RTP audio format 106
  122. Found RTP audio format 9
  123. Found RTP audio format 3
  124. Found RTP audio format 111
  125. Found RTP audio format 0
  126. Found RTP audio format 8
  127. Found RTP audio format 97
  128. Found RTP audio format 110
  129. Found RTP audio format 112
  130. Found RTP audio format 98
  131. Found RTP audio format 101
  132. Found RTP audio format 100
  133. Found RTP audio format 99
  134. Found RTP audio format 102
  135. Found audio description format opus for ID 106
  136. Found audio description format speex for ID 111
  137. Found audio description format iLBC for ID 97
  138. Found audio description format speex for ID 110
  139. Found audio description format speex for ID 112
  140. Found unknown media description format telephone-event for ID 98
  141. Found audio description format telephone-event for ID 101
  142. Found unknown media description format telephone-event for ID 100
  143. Found unknown media description format telephone-event for ID 99
  144. Found audio description format G726-32 for ID 102
  145. Capabilities: us - (ulaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|g726|opus|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw)
  146. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  147.        > 0x7f186801adf0 -- Strict RTP learning after remote address set to: 192.168.0.94:8000
  148. Peer audio RTP is at port 192.168.0.94:8000
  149. Looking for 100 in from-internal (domain 40.122.50.92)
  150. sip_route_dump: route/path hop: <sip:6001@192.168.0.94:5061;transport=UDP>
  151.  
  152. <--- Transmitting (NAT) to 97.122.234.169:5061 --->
  153. SIP/2.0 100 Trying
  154. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  155. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  156. To: <sip:100@40.122.50.92;transport=UDP>
  157. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  158. CSeq: 2 INVITE
  159. Server: Asterisk PBX 16.5.0
  160. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  161. Supported: replaces, timer
  162. Contact: <sip:100@10.0.0.4:5060>
  163. Content-Length: 0
  164.  
  165.  
  166. <------------>
  167.     -- Executing [100@from-internal:1] Answer("SIP/6001-00000004", "") in new stack
  168. Audio is at 37540
  169. Adding codec ulaw to SDP
  170. Adding non-codec 0x1 (telephone-event) to SDP
  171.  
  172. <--- Reliably Transmitting (NAT) to 97.122.234.169:5061 --->
  173. SIP/2.0 200 OK
  174. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  175. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  176. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  177. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  178. CSeq: 2 INVITE
  179. Server: Asterisk PBX 16.5.0
  180. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  181. Supported: replaces, timer
  182. Contact: <sip:100@10.0.0.4:5060>
  183. Content-Type: application/sdp
  184. Content-Length: 231
  185.  
  186. v=0
  187. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  188. s=Asterisk PBX 16.5.0
  189. c=IN IP4 10.0.0.4
  190. t=0 0
  191. m=audio 37540 RTP/AVP 0 101
  192. a=rtpmap:0 PCMU/8000
  193. a=rtpmap:101 telephone-event/8000
  194. a=fmtp:101 0-16
  195. a=maxptime:150
  196. a=sendrecv
  197.  
  198. <------------>
  199. Retransmitting #1 (NAT) to 97.122.234.169:5061:
  200. SIP/2.0 200 OK
  201. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  202. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  203. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  204. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  205. CSeq: 2 INVITE
  206. Server: Asterisk PBX 16.5.0
  207. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  208. Supported: replaces, timer
  209. Contact: <sip:100@10.0.0.4:5060>
  210. Content-Type: application/sdp
  211. Content-Length: 231
  212.  
  213. v=0
  214. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  215. s=Asterisk PBX 16.5.0
  216. c=IN IP4 10.0.0.4
  217. t=0 0
  218. m=audio 37540 RTP/AVP 0 101
  219. a=rtpmap:0 PCMU/8000
  220. a=rtpmap:101 telephone-event/8000
  221. a=fmtp:101 0-16
  222. a=maxptime:150
  223. a=sendrecv
  224.  
  225. ---
  226.     -- Executing [100@from-internal:2] Wait("SIP/6001-00000004", "1") in new stack
  227. Retransmitting #2 (NAT) to 97.122.234.169:5061:
  228. SIP/2.0 200 OK
  229. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  230. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  231. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  232. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  233. CSeq: 2 INVITE
  234. Server: Asterisk PBX 16.5.0
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  236. Supported: replaces, timer
  237. Contact: <sip:100@10.0.0.4:5060>
  238. Content-Type: application/sdp
  239. Content-Length: 231
  240.  
  241. v=0
  242. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  243. s=Asterisk PBX 16.5.0
  244. c=IN IP4 10.0.0.4
  245. t=0 0
  246. m=audio 37540 RTP/AVP 0 101
  247. a=rtpmap:0 PCMU/8000
  248. a=rtpmap:101 telephone-event/8000
  249. a=fmtp:101 0-16
  250. a=maxptime:150
  251. a=sendrecv
  252.  
  253. ---
  254.     -- Executing [100@from-internal:3] Playback("SIP/6001-00000004", "hello-world") in new stack
  255.     -- <SIP/6001-00000004> Playing 'hello-world.ulaw' (language 'en')
  256.     -- Executing [100@from-internal:4] Hangup("SIP/6001-00000004", "") in new stack
  257.   == Spawn extension (from-internal, 100, 4) exited non-zero on 'SIP/6001-00000004'
  258. Scheduling destruction of SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' in 32000 ms (Method: INVITE)
  259. Retransmitting #3 (NAT) to 97.122.234.169:5061:
  260. SIP/2.0 200 OK
  261. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  262. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  263. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  264. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  265. CSeq: 2 INVITE
  266. Server: Asterisk PBX 16.5.0
  267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  268. Supported: replaces, timer
  269. Contact: <sip:100@10.0.0.4:5060>
  270. Content-Type: application/sdp
  271. Content-Length: 231
  272.  
  273. v=0
  274. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  275. s=Asterisk PBX 16.5.0
  276. c=IN IP4 10.0.0.4
  277. t=0 0
  278. m=audio 37540 RTP/AVP 0 101
  279. a=rtpmap:0 PCMU/8000
  280. a=rtpmap:101 telephone-event/8000
  281. a=fmtp:101 0-16
  282. a=maxptime:150
  283. a=sendrecv
  284.  
  285. ---
  286. Retransmitting #4 (NAT) to 97.122.234.169:5061:
  287. SIP/2.0 200 OK
  288. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  289. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  290. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  291. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  292. CSeq: 2 INVITE
  293. Server: Asterisk PBX 16.5.0
  294. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  295. Supported: replaces, timer
  296. Contact: <sip:100@10.0.0.4:5060>
  297. Content-Type: application/sdp
  298. Content-Length: 231
  299.  
  300. v=0
  301. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  302. s=Asterisk PBX 16.5.0
  303. c=IN IP4 10.0.0.4
  304. t=0 0
  305. m=audio 37540 RTP/AVP 0 101
  306. a=rtpmap:0 PCMU/8000
  307. a=rtpmap:101 telephone-event/8000
  308. a=fmtp:101 0-16
  309. a=maxptime:150
  310. a=sendrecv
  311.  
  312. ---
  313.  
  314. <--- SIP read from UDP:97.122.234.169:5061 --->
  315. REGISTER sip:40.122.50.92;transport=UDP SIP/2.0
  316. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---625ae9cff6f1b119;rport
  317. Max-Forwards: 70
  318. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>
  319. To: <sip:6001@40.122.50.92;transport=UDP>
  320. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  321. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  322. CSeq: 13141 REGISTER
  323. Expires: 60
  324. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  325. User-Agent: Z 5.2.28 rv2.8.114
  326. Authorization: Digest username="6001",realm="asterisk",nonce="51ebdf5a",uri="sip:40.122.50.92;transport=UDP",response="1d0a98f7f5bff64fbb7f9359af34e48d",algorithm=MD5
  327. Allow-Events: presence, kpml, talk
  328. Content-Length: 0
  329.  
  330. <------------->
  331. --- (14 headers 0 lines) ---
  332. Sending to 97.122.234.169:5061 (NAT)
  333. Sending to 97.122.234.169:5061 (NAT)
  334.  
  335. <--- Transmitting (NAT) to 97.122.234.169:5061 --->
  336. SIP/2.0 401 Unauthorized
  337. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---625ae9cff6f1b119;received=97.122.234.169;rport=5061
  338. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  339. To: <sip:6001@40.122.50.92;transport=UDP>;tag=as38a18719
  340. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  341. CSeq: 13141 REGISTER
  342. Server: Asterisk PBX 16.5.0
  343. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  344. Supported: replaces, timer
  345. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67f14424"
  346. Content-Length: 0
  347.  
  348.  
  349. <------------>
  350. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  351.  
  352. <--- SIP read from UDP:97.122.234.169:5061 --->
  353. REGISTER sip:40.122.50.92;transport=UDP SIP/2.0
  354. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---f89f22257a530858;rport
  355. Max-Forwards: 70
  356. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>
  357. To: <sip:6001@40.122.50.92;transport=UDP>
  358. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  359. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  360. CSeq: 13142 REGISTER
  361. Expires: 60
  362. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  363. User-Agent: Z 5.2.28 rv2.8.114
  364. Authorization: Digest username="6001",realm="asterisk",nonce="67f14424",uri="sip:40.122.50.92;transport=UDP",response="d2df9ffd1eea02262bdaf940de845411",algorithm=MD5
  365. Allow-Events: presence, kpml, talk
  366. Content-Length: 0
  367.  
  368. <------------->
  369. --- (14 headers 0 lines) ---
  370. Sending to 97.122.234.169:5061 (NAT)
  371.  
  372. <--- Transmitting (NAT) to 97.122.234.169:5061 --->
  373. SIP/2.0 200 OK
  374. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---f89f22257a530858;received=97.122.234.169;rport=5061
  375. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  376. To: <sip:6001@40.122.50.92;transport=UDP>;tag=as38a18719
  377. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  378. CSeq: 13142 REGISTER
  379. Server: Asterisk PBX 16.5.0
  380. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  381. Supported: replaces, timer
  382. Expires: 60
  383. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>;expires=60
  384. Date: Mon, 26 Aug 2019 21:18:30 GMT
  385. Content-Length: 0
  386.  
  387.  
  388. <------------>
  389. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  390. Retransmitting #5 (NAT) to 97.122.234.169:5061:
  391. SIP/2.0 200 OK
  392. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  393. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  394. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  395. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  396. CSeq: 2 INVITE
  397. Server: Asterisk PBX 16.5.0
  398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  399. Supported: replaces, timer
  400. Contact: <sip:100@10.0.0.4:5060>
  401. Content-Type: application/sdp
  402. Content-Length: 231
  403.  
  404. v=0
  405. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  406. s=Asterisk PBX 16.5.0
  407. c=IN IP4 10.0.0.4
  408. t=0 0
  409. m=audio 37540 RTP/AVP 0 101
  410. a=rtpmap:0 PCMU/8000
  411. a=rtpmap:101 telephone-event/8000
  412. a=fmtp:101 0-16
  413. a=maxptime:150
  414. a=sendrecv
  415.  
  416. ---
  417.  
  418. <--- SIP read from UDP:97.122.234.169:5061 --->
  419.  
  420.  
  421. <------------->
  422. Retransmitting #6 (NAT) to 97.122.234.169:5061:
  423. SIP/2.0 200 OK
  424. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  425. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  426. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  427. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  428. CSeq: 2 INVITE
  429. Server: Asterisk PBX 16.5.0
  430. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  431. Supported: replaces, timer
  432. Contact: <sip:100@10.0.0.4:5060>
  433. Content-Type: application/sdp
  434. Content-Length: 231
  435.  
  436. v=0
  437. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  438. s=Asterisk PBX 16.5.0
  439. c=IN IP4 10.0.0.4
  440. t=0 0
  441. m=audio 37540 RTP/AVP 0 101
  442. a=rtpmap:0 PCMU/8000
  443. a=rtpmap:101 telephone-event/8000
  444. a=fmtp:101 0-16
  445. a=maxptime:150
  446. a=sendrecv
  447.  
  448. ---
  449. Retransmitting #7 (NAT) to 97.122.234.169:5061:
  450. SIP/2.0 200 OK
  451. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  452. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  453. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  454. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  455. CSeq: 2 INVITE
  456. Server: Asterisk PBX 16.5.0
  457. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  458. Supported: replaces, timer
  459. Contact: <sip:100@10.0.0.4:5060>
  460. Content-Type: application/sdp
  461. Content-Length: 231
  462.  
  463. v=0
  464. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  465. s=Asterisk PBX 16.5.0
  466. c=IN IP4 10.0.0.4
  467. t=0 0
  468. m=audio 37540 RTP/AVP 0 101
  469. a=rtpmap:0 PCMU/8000
  470. a=rtpmap:101 telephone-event/8000
  471. a=fmtp:101 0-16
  472. a=maxptime:150
  473. a=sendrecv
  474.  
  475. ---
  476. Retransmitting #8 (NAT) to 97.122.234.169:5061:
  477. SIP/2.0 200 OK
  478. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  479. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  480. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  481. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  482. CSeq: 2 INVITE
  483. Server: Asterisk PBX 16.5.0
  484. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  485. Supported: replaces, timer
  486. Contact: <sip:100@10.0.0.4:5060>
  487. Content-Type: application/sdp
  488. Content-Length: 231
  489.  
  490. v=0
  491. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  492. s=Asterisk PBX 16.5.0
  493. c=IN IP4 10.0.0.4
  494. t=0 0
  495. m=audio 37540 RTP/AVP 0 101
  496. a=rtpmap:0 PCMU/8000
  497. a=rtpmap:101 telephone-event/8000
  498. a=fmtp:101 0-16
  499. a=maxptime:150
  500. a=sendrecv
  501.  
  502. ---
  503. Retransmitting #9 (NAT) to 97.122.234.169:5061:
  504. SIP/2.0 200 OK
  505. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  506. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  507. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  508. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  509. CSeq: 2 INVITE
  510. Server: Asterisk PBX 16.5.0
  511. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  512. Supported: replaces, timer
  513. Contact: <sip:100@10.0.0.4:5060>
  514. Content-Type: application/sdp
  515. Content-Length: 231
  516.  
  517. v=0
  518. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  519. s=Asterisk PBX 16.5.0
  520. c=IN IP4 10.0.0.4
  521. t=0 0
  522. m=audio 37540 RTP/AVP 0 101
  523. a=rtpmap:0 PCMU/8000
  524. a=rtpmap:101 telephone-event/8000
  525. a=fmtp:101 0-16
  526. a=maxptime:150
  527. a=sendrecv
  528.  
  529. ---
  530. Retransmitting #10 (NAT) to 97.122.234.169:5061:
  531. SIP/2.0 200 OK
  532. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=97.122.234.169;rport=5061
  533. From: <sip:6001@40.122.50.92;transport=UDP>;tag=6503de27
  534. To: <sip:100@40.122.50.92;transport=UDP>;tag=as3a9ae648
  535. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  536. CSeq: 2 INVITE
  537. Server: Asterisk PBX 16.5.0
  538. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  539. Supported: replaces, timer
  540. Contact: <sip:100@10.0.0.4:5060>
  541. Content-Type: application/sdp
  542. Content-Length: 231
  543.  
  544. v=0
  545. o=root 1150555715 1150555715 IN IP4 10.0.0.4
  546. s=Asterisk PBX 16.5.0
  547. c=IN IP4 10.0.0.4
  548. t=0 0
  549. m=audio 37540 RTP/AVP 0 101
  550. a=rtpmap:0 PCMU/8000
  551. a=rtpmap:101 telephone-event/8000
  552. a=fmtp:101 0-16
  553. a=maxptime:150
  554. a=sendrecv
  555.  
  556. ---
  557. [Aug 26 21:18:54] WARNING[1984]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission kv40PFslKyjCAY7ZiEL9kA.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  558. Packet timed out after 32000ms with no response
  559. Really destroying SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' Method: INVITE
  560. Really destroying SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' Method: REGISTER
  561.  
  562. <--- SIP read from UDP:97.122.234.169:5061 --->
  563.  
  564.  
  565. <------------->
  566.  
  567. <--- SIP read from UDP:97.122.234.169:5061 --->
  568. REGISTER sip:40.122.50.92;transport=UDP SIP/2.0
  569. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8ec7eeb6edc0735a;rport
  570. Max-Forwards: 70
  571. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>
  572. To: <sip:6001@40.122.50.92;transport=UDP>
  573. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  574. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  575. CSeq: 13143 REGISTER
  576. Expires: 60
  577. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  578. User-Agent: Z 5.2.28 rv2.8.114
  579. Authorization: Digest username="6001",realm="asterisk",nonce="67f14424",uri="sip:40.122.50.92;transport=UDP",response="d2df9ffd1eea02262bdaf940de845411",algorithm=MD5
  580. Allow-Events: presence, kpml, talk
  581. Content-Length: 0
  582.  
  583. <------------->
  584. --- (14 headers 0 lines) ---
  585. Sending to 97.122.234.169:5061 (NAT)
  586. Sending to 97.122.234.169:5061 (NAT)
  587.  
  588. <--- Transmitting (NAT) to 97.122.234.169:5061 --->
  589. SIP/2.0 401 Unauthorized
  590. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---8ec7eeb6edc0735a;received=97.122.234.169;rport=5061
  591. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  592. To: <sip:6001@40.122.50.92;transport=UDP>;tag=as1613b627
  593. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  594. CSeq: 13143 REGISTER
  595. Server: Asterisk PBX 16.5.0
  596. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  597. Supported: replaces, timer
  598. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ab2c4b3"
  599. Content-Length: 0
  600.  
  601.  
  602. <------------>
  603. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  604.  
  605. <--- SIP read from UDP:97.122.234.169:5061 --->
  606. REGISTER sip:40.122.50.92;transport=UDP SIP/2.0
  607. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---dbf8c9aa7345882c;rport
  608. Max-Forwards: 70
  609. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>
  610. To: <sip:6001@40.122.50.92;transport=UDP>
  611. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  612. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  613. CSeq: 13144 REGISTER
  614. Expires: 60
  615. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  616. User-Agent: Z 5.2.28 rv2.8.114
  617. Authorization: Digest username="6001",realm="asterisk",nonce="3ab2c4b3",uri="sip:40.122.50.92;transport=UDP",response="2830ebbdd36787a5ef0e80ef4dd50c96",algorithm=MD5
  618. Allow-Events: presence, kpml, talk
  619. Content-Length: 0
  620.  
  621. <------------->
  622. --- (14 headers 0 lines) ---
  623. Sending to 97.122.234.169:5061 (NAT)
  624.  
  625. <--- Transmitting (NAT) to 97.122.234.169:5061 --->
  626. SIP/2.0 200 OK
  627. Via: SIP/2.0/UDP 192.168.0.94:5061;branch=z9hG4bK-524287-1---dbf8c9aa7345882c;received=97.122.234.169;rport=5061
  628. From: <sip:6001@40.122.50.92;transport=UDP>;tag=a14d1338
  629. To: <sip:6001@40.122.50.92;transport=UDP>;tag=as1613b627
  630. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  631. CSeq: 13144 REGISTER
  632. Server: Asterisk PBX 16.5.0
  633. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  634. Supported: replaces, timer
  635. Expires: 60
  636. Contact: <sip:6001@107.2.246.165:5061;rinstance=019a188f09cfde6b;transport=UDP>;expires=60
  637. Date: Mon, 26 Aug 2019 21:19:24 GMT
  638. Content-Length: 0
  639.  
  640.  
  641. <------------>
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