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  1.  
  2. <--- SIP read from UDP:my_ip:5061 --->
  3. INVITE sip:100@server_ip;transport=UDP SIP/2.0
  4. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;rport
  5. Max-Forwards: 70
  6. Contact: <sip:6001@client_subnet:5061;transport=UDP>
  7. To: <sip:100@server_ip;transport=UDP>
  8. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  9. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  10. CSeq: 1 INVITE
  11. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  12. Content-Type: application/sdp
  13. User-Agent: Z 5.2.28 rv2.8.114
  14. Allow-Events: presence, kpml, talk
  15. Content-Length: 608
  16.  
  17. v=0
  18. o=Z 401744695 0 IN IP4 client_subnet
  19. s=Z
  20. c=IN IP4 client_subnet
  21. t=0 0
  22. m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
  23. a=rtpmap:106 opus/48000/2
  24. a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
  25. a=rtpmap:111 speex/16000
  26. a=rtpmap:97 iLBC/8000
  27. a=fmtp:97 mode=20
  28. a=rtpmap:110 speex/8000
  29. a=rtpmap:112 speex/32000
  30. a=rtpmap:98 telephone-event/48000
  31. a=fmtp:98 0-16
  32. a=rtpmap:101 telephone-event/8000
  33. a=fmtp:101 0-16
  34. a=rtpmap:100 telephone-event/16000
  35. a=fmtp:100 0-16
  36. a=rtpmap:99 telephone-event/32000
  37. a=fmtp:99 0-16
  38. a=rtpmap:102 G726-32/8000
  39. a=sendrecv
  40. <------------->
  41. --- (13 headers 23 lines) ---
  42. Sending to my_ip:5061 (NAT)
  43. Sending to my_ip:5061 (NAT)
  44. Using INVITE request as basis request - kv40PFslKyjCAY7ZiEL9kA..
  45. Found peer '6001' for '6001' from my_ip:5061
  46.  
  47. <--- Reliably Transmitting (NAT) to my_ip:5061 --->
  48. SIP/2.0 401 Unauthorized
  49. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;received=my_ip;rport=5061
  50. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  51. To: <sip:100@server_ip;transport=UDP>;tag=as191bd6c4
  52. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  53. CSeq: 1 INVITE
  54. Server: Asterisk PBX 16.5.0
  55. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  56. Supported: replaces, timer
  57. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02597d71"
  58. Content-Length: 0
  59.  
  60.  
  61. <------------>
  62. Scheduling destruction of SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' in 32000 ms (Method: INVITE)
  63.  
  64. <--- SIP read from UDP:my_ip:5061 --->
  65. ACK sip:100@server_ip;transport=UDP SIP/2.0
  66. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---efed32d92b6ddd24;rport
  67. Max-Forwards: 70
  68. To: <sip:100@server_ip;transport=UDP>;tag=as191bd6c4
  69. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  70. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  71. CSeq: 1 ACK
  72. Content-Length: 0
  73.  
  74. <------------->
  75. --- (8 headers 0 lines) ---
  76.  
  77. <--- SIP read from UDP:my_ip:5061 --->
  78. INVITE sip:100@server_ip;transport=UDP SIP/2.0
  79. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;rport
  80. Max-Forwards: 70
  81. Contact: <sip:6001@client_subnet:5061;transport=UDP>
  82. To: <sip:100@server_ip;transport=UDP>
  83. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  84. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  85. CSeq: 2 INVITE
  86. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  87. Content-Type: application/sdp
  88. User-Agent: Z 5.2.28 rv2.8.114
  89. Authorization: Digest username="6001",realm="asterisk",nonce="02597d71",uri="sip:100@server_ip;transport=UDP",response="1858388598874fd6c70b5015c49289f0",algorithm=MD5
  90. Allow-Events: presence, kpml, talk
  91. Content-Length: 608
  92.  
  93. v=0
  94. o=Z 401744695 0 IN IP4 client_subnet
  95. s=Z
  96. c=IN IP4 client_subnet
  97. t=0 0
  98. m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
  99. a=rtpmap:106 opus/48000/2
  100. a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
  101. a=rtpmap:111 speex/16000
  102. a=rtpmap:97 iLBC/8000
  103. a=fmtp:97 mode=20
  104. a=rtpmap:110 speex/8000
  105. a=rtpmap:112 speex/32000
  106. a=rtpmap:98 telephone-event/48000
  107. a=fmtp:98 0-16
  108. a=rtpmap:101 telephone-event/8000
  109. a=fmtp:101 0-16
  110. a=rtpmap:100 telephone-event/16000
  111. a=fmtp:100 0-16
  112. a=rtpmap:99 telephone-event/32000
  113. a=fmtp:99 0-16
  114. a=rtpmap:102 G726-32/8000
  115. a=sendrecv
  116. <------------->
  117. --- (14 headers 23 lines) ---
  118. Sending to my_ip:5061 (NAT)
  119. Using INVITE request as basis request - kv40PFslKyjCAY7ZiEL9kA..
  120. Found peer '6001' for '6001' from my_ip:5061
  121.   == Using SIP RTP CoS mark 5
  122. Found RTP audio format 106
  123. Found RTP audio format 9
  124. Found RTP audio format 3
  125. Found RTP audio format 111
  126. Found RTP audio format 0
  127. Found RTP audio format 8
  128. Found RTP audio format 97
  129. Found RTP audio format 110
  130. Found RTP audio format 112
  131. Found RTP audio format 98
  132. Found RTP audio format 101
  133. Found RTP audio format 100
  134. Found RTP audio format 99
  135. Found RTP audio format 102
  136. Found audio description format opus for ID 106
  137. Found audio description format speex for ID 111
  138. Found audio description format iLBC for ID 97
  139. Found audio description format speex for ID 110
  140. Found audio description format speex for ID 112
  141. Found unknown media description format telephone-event for ID 98
  142. Found audio description format telephone-event for ID 101
  143. Found unknown media description format telephone-event for ID 100
  144. Found unknown media description format telephone-event for ID 99
  145. Found audio description format G726-32 for ID 102
  146. Capabilities: us - (ulaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|g726|opus|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw)
  147. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  148.        > 0x7f186801adf0 -- Strict RTP learning after remote address set to: client_subnet:8000
  149. Peer audio RTP is at port client_subnet:8000
  150. Looking for 100 in from-internal (domain server_ip)
  151. sip_route_dump: route/path hop: <sip:6001@client_subnet:5061;transport=UDP>
  152.  
  153. <--- Transmitting (NAT) to my_ip:5061 --->
  154. SIP/2.0 100 Trying
  155. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  156. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  157. To: <sip:100@server_ip;transport=UDP>
  158. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  159. CSeq: 2 INVITE
  160. Server: Asterisk PBX 16.5.0
  161. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  162. Supported: replaces, timer
  163. Contact: <sip:100@server_subnet:5060>
  164. Content-Length: 0
  165.  
  166.  
  167. <------------>
  168.     -- Executing [100@from-internal:1] Answer("SIP/6001-00000004", "") in new stack
  169. Audio is at 37540
  170. Adding codec ulaw to SDP
  171. Adding non-codec 0x1 (telephone-event) to SDP
  172.  
  173. <--- Reliably Transmitting (NAT) to my_ip:5061 --->
  174. SIP/2.0 200 OK
  175. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  176. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  177. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  178. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  179. CSeq: 2 INVITE
  180. Server: Asterisk PBX 16.5.0
  181. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  182. Supported: replaces, timer
  183. Contact: <sip:100@server_subnet:5060>
  184. Content-Type: application/sdp
  185. Content-Length: 231
  186.  
  187. v=0
  188. o=root 1150555715 1150555715 IN IP4 server_subnet
  189. s=Asterisk PBX 16.5.0
  190. c=IN IP4 server_subnet
  191. t=0 0
  192. m=audio 37540 RTP/AVP 0 101
  193. a=rtpmap:0 PCMU/8000
  194. a=rtpmap:101 telephone-event/8000
  195. a=fmtp:101 0-16
  196. a=maxptime:150
  197. a=sendrecv
  198.  
  199. <------------>
  200. Retransmitting #1 (NAT) to my_ip:5061:
  201. SIP/2.0 200 OK
  202. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  203. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  204. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  205. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  206. CSeq: 2 INVITE
  207. Server: Asterisk PBX 16.5.0
  208. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  209. Supported: replaces, timer
  210. Contact: <sip:100@server_subnet:5060>
  211. Content-Type: application/sdp
  212. Content-Length: 231
  213.  
  214. v=0
  215. o=root 1150555715 1150555715 IN IP4 server_subnet
  216. s=Asterisk PBX 16.5.0
  217. c=IN IP4 server_subnet
  218. t=0 0
  219. m=audio 37540 RTP/AVP 0 101
  220. a=rtpmap:0 PCMU/8000
  221. a=rtpmap:101 telephone-event/8000
  222. a=fmtp:101 0-16
  223. a=maxptime:150
  224. a=sendrecv
  225.  
  226. ---
  227.     -- Executing [100@from-internal:2] Wait("SIP/6001-00000004", "1") in new stack
  228. Retransmitting #2 (NAT) to my_ip:5061:
  229. SIP/2.0 200 OK
  230. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  231. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  232. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  233. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  234. CSeq: 2 INVITE
  235. Server: Asterisk PBX 16.5.0
  236. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  237. Supported: replaces, timer
  238. Contact: <sip:100@server_subnet:5060>
  239. Content-Type: application/sdp
  240. Content-Length: 231
  241.  
  242. v=0
  243. o=root 1150555715 1150555715 IN IP4 server_subnet
  244. s=Asterisk PBX 16.5.0
  245. c=IN IP4 server_subnet
  246. t=0 0
  247. m=audio 37540 RTP/AVP 0 101
  248. a=rtpmap:0 PCMU/8000
  249. a=rtpmap:101 telephone-event/8000
  250. a=fmtp:101 0-16
  251. a=maxptime:150
  252. a=sendrecv
  253.  
  254. ---
  255.     -- Executing [100@from-internal:3] Playback("SIP/6001-00000004", "hello-world") in new stack
  256.     -- <SIP/6001-00000004> Playing 'hello-world.ulaw' (language 'en')
  257.     -- Executing [100@from-internal:4] Hangup("SIP/6001-00000004", "") in new stack
  258.   == Spawn extension (from-internal, 100, 4) exited non-zero on 'SIP/6001-00000004'
  259. Scheduling destruction of SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' in 32000 ms (Method: INVITE)
  260. Retransmitting #3 (NAT) to my_ip:5061:
  261. SIP/2.0 200 OK
  262. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  263. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  264. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  265. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  266. CSeq: 2 INVITE
  267. Server: Asterisk PBX 16.5.0
  268. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  269. Supported: replaces, timer
  270. Contact: <sip:100@server_subnet:5060>
  271. Content-Type: application/sdp
  272. Content-Length: 231
  273.  
  274. v=0
  275. o=root 1150555715 1150555715 IN IP4 server_subnet
  276. s=Asterisk PBX 16.5.0
  277. c=IN IP4 server_subnet
  278. t=0 0
  279. m=audio 37540 RTP/AVP 0 101
  280. a=rtpmap:0 PCMU/8000
  281. a=rtpmap:101 telephone-event/8000
  282. a=fmtp:101 0-16
  283. a=maxptime:150
  284. a=sendrecv
  285.  
  286. ---
  287. Retransmitting #4 (NAT) to my_ip:5061:
  288. SIP/2.0 200 OK
  289. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  290. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  291. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  292. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  293. CSeq: 2 INVITE
  294. Server: Asterisk PBX 16.5.0
  295. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  296. Supported: replaces, timer
  297. Contact: <sip:100@server_subnet:5060>
  298. Content-Type: application/sdp
  299. Content-Length: 231
  300.  
  301. v=0
  302. o=root 1150555715 1150555715 IN IP4 server_subnet
  303. s=Asterisk PBX 16.5.0
  304. c=IN IP4 server_subnet
  305. t=0 0
  306. m=audio 37540 RTP/AVP 0 101
  307. a=rtpmap:0 PCMU/8000
  308. a=rtpmap:101 telephone-event/8000
  309. a=fmtp:101 0-16
  310. a=maxptime:150
  311. a=sendrecv
  312.  
  313. ---
  314.  
  315. <--- SIP read from UDP:my_ip:5061 --->
  316. REGISTER sip:server_ip;transport=UDP SIP/2.0
  317. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---625ae9cff6f1b119;rport
  318. Max-Forwards: 70
  319. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>
  320. To: <sip:6001@server_ip;transport=UDP>
  321. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  322. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  323. CSeq: 13141 REGISTER
  324. Expires: 60
  325. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  326. User-Agent: Z 5.2.28 rv2.8.114
  327. Authorization: Digest username="6001",realm="asterisk",nonce="51ebdf5a",uri="sip:server_ip;transport=UDP",response="1d0a98f7f5bff64fbb7f9359af34e48d",algorithm=MD5
  328. Allow-Events: presence, kpml, talk
  329. Content-Length: 0
  330.  
  331. <------------->
  332. --- (14 headers 0 lines) ---
  333. Sending to my_ip:5061 (NAT)
  334. Sending to my_ip:5061 (NAT)
  335.  
  336. <--- Transmitting (NAT) to my_ip:5061 --->
  337. SIP/2.0 401 Unauthorized
  338. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---625ae9cff6f1b119;received=my_ip;rport=5061
  339. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  340. To: <sip:6001@server_ip;transport=UDP>;tag=as38a18719
  341. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  342. CSeq: 13141 REGISTER
  343. Server: Asterisk PBX 16.5.0
  344. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  345. Supported: replaces, timer
  346. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67f14424"
  347. Content-Length: 0
  348.  
  349.  
  350. <------------>
  351. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  352.  
  353. <--- SIP read from UDP:my_ip:5061 --->
  354. REGISTER sip:server_ip;transport=UDP SIP/2.0
  355. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---f89f22257a530858;rport
  356. Max-Forwards: 70
  357. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>
  358. To: <sip:6001@server_ip;transport=UDP>
  359. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  360. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  361. CSeq: 13142 REGISTER
  362. Expires: 60
  363. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  364. User-Agent: Z 5.2.28 rv2.8.114
  365. Authorization: Digest username="6001",realm="asterisk",nonce="67f14424",uri="sip:server_ip;transport=UDP",response="d2df9ffd1eea02262bdaf940de845411",algorithm=MD5
  366. Allow-Events: presence, kpml, talk
  367. Content-Length: 0
  368.  
  369. <------------->
  370. --- (14 headers 0 lines) ---
  371. Sending to my_ip:5061 (NAT)
  372.  
  373. <--- Transmitting (NAT) to my_ip:5061 --->
  374. SIP/2.0 200 OK
  375. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---f89f22257a530858;received=my_ip;rport=5061
  376. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  377. To: <sip:6001@server_ip;transport=UDP>;tag=as38a18719
  378. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  379. CSeq: 13142 REGISTER
  380. Server: Asterisk PBX 16.5.0
  381. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  382. Supported: replaces, timer
  383. Expires: 60
  384. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>;expires=60
  385. Date: Mon, 26 Aug 2019 21:18:30 GMT
  386. Content-Length: 0
  387.  
  388.  
  389. <------------>
  390. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  391. Retransmitting #5 (NAT) to my_ip:5061:
  392. SIP/2.0 200 OK
  393. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  394. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  395. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  396. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  397. CSeq: 2 INVITE
  398. Server: Asterisk PBX 16.5.0
  399. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  400. Supported: replaces, timer
  401. Contact: <sip:100@server_subnet:5060>
  402. Content-Type: application/sdp
  403. Content-Length: 231
  404.  
  405. v=0
  406. o=root 1150555715 1150555715 IN IP4 server_subnet
  407. s=Asterisk PBX 16.5.0
  408. c=IN IP4 server_subnet
  409. t=0 0
  410. m=audio 37540 RTP/AVP 0 101
  411. a=rtpmap:0 PCMU/8000
  412. a=rtpmap:101 telephone-event/8000
  413. a=fmtp:101 0-16
  414. a=maxptime:150
  415. a=sendrecv
  416.  
  417. ---
  418.  
  419. <--- SIP read from UDP:my_ip:5061 --->
  420.  
  421.  
  422. <------------->
  423. Retransmitting #6 (NAT) to my_ip:5061:
  424. SIP/2.0 200 OK
  425. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  426. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  427. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  428. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  429. CSeq: 2 INVITE
  430. Server: Asterisk PBX 16.5.0
  431. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  432. Supported: replaces, timer
  433. Contact: <sip:100@server_subnet:5060>
  434. Content-Type: application/sdp
  435. Content-Length: 231
  436.  
  437. v=0
  438. o=root 1150555715 1150555715 IN IP4 server_subnet
  439. s=Asterisk PBX 16.5.0
  440. c=IN IP4 server_subnet
  441. t=0 0
  442. m=audio 37540 RTP/AVP 0 101
  443. a=rtpmap:0 PCMU/8000
  444. a=rtpmap:101 telephone-event/8000
  445. a=fmtp:101 0-16
  446. a=maxptime:150
  447. a=sendrecv
  448.  
  449. ---
  450. Retransmitting #7 (NAT) to my_ip:5061:
  451. SIP/2.0 200 OK
  452. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  453. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  454. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  455. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  456. CSeq: 2 INVITE
  457. Server: Asterisk PBX 16.5.0
  458. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  459. Supported: replaces, timer
  460. Contact: <sip:100@server_subnet:5060>
  461. Content-Type: application/sdp
  462. Content-Length: 231
  463.  
  464. v=0
  465. o=root 1150555715 1150555715 IN IP4 server_subnet
  466. s=Asterisk PBX 16.5.0
  467. c=IN IP4 server_subnet
  468. t=0 0
  469. m=audio 37540 RTP/AVP 0 101
  470. a=rtpmap:0 PCMU/8000
  471. a=rtpmap:101 telephone-event/8000
  472. a=fmtp:101 0-16
  473. a=maxptime:150
  474. a=sendrecv
  475.  
  476. ---
  477. Retransmitting #8 (NAT) to my_ip:5061:
  478. SIP/2.0 200 OK
  479. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  480. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  481. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  482. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  483. CSeq: 2 INVITE
  484. Server: Asterisk PBX 16.5.0
  485. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  486. Supported: replaces, timer
  487. Contact: <sip:100@server_subnet:5060>
  488. Content-Type: application/sdp
  489. Content-Length: 231
  490.  
  491. v=0
  492. o=root 1150555715 1150555715 IN IP4 server_subnet
  493. s=Asterisk PBX 16.5.0
  494. c=IN IP4 server_subnet
  495. t=0 0
  496. m=audio 37540 RTP/AVP 0 101
  497. a=rtpmap:0 PCMU/8000
  498. a=rtpmap:101 telephone-event/8000
  499. a=fmtp:101 0-16
  500. a=maxptime:150
  501. a=sendrecv
  502.  
  503. ---
  504. Retransmitting #9 (NAT) to my_ip:5061:
  505. SIP/2.0 200 OK
  506. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  507. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  508. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  509. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  510. CSeq: 2 INVITE
  511. Server: Asterisk PBX 16.5.0
  512. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  513. Supported: replaces, timer
  514. Contact: <sip:100@server_subnet:5060>
  515. Content-Type: application/sdp
  516. Content-Length: 231
  517.  
  518. v=0
  519. o=root 1150555715 1150555715 IN IP4 server_subnet
  520. s=Asterisk PBX 16.5.0
  521. c=IN IP4 server_subnet
  522. t=0 0
  523. m=audio 37540 RTP/AVP 0 101
  524. a=rtpmap:0 PCMU/8000
  525. a=rtpmap:101 telephone-event/8000
  526. a=fmtp:101 0-16
  527. a=maxptime:150
  528. a=sendrecv
  529.  
  530. ---
  531. Retransmitting #10 (NAT) to my_ip:5061:
  532. SIP/2.0 200 OK
  533. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8b81bcac6012a919;received=my_ip;rport=5061
  534. From: <sip:6001@server_ip;transport=UDP>;tag=6503de27
  535. To: <sip:100@server_ip;transport=UDP>;tag=as3a9ae648
  536. Call-ID: kv40PFslKyjCAY7ZiEL9kA..
  537. CSeq: 2 INVITE
  538. Server: Asterisk PBX 16.5.0
  539. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  540. Supported: replaces, timer
  541. Contact: <sip:100@server_subnet:5060>
  542. Content-Type: application/sdp
  543. Content-Length: 231
  544.  
  545. v=0
  546. o=root 1150555715 1150555715 IN IP4 server_subnet
  547. s=Asterisk PBX 16.5.0
  548. c=IN IP4 server_subnet
  549. t=0 0
  550. m=audio 37540 RTP/AVP 0 101
  551. a=rtpmap:0 PCMU/8000
  552. a=rtpmap:101 telephone-event/8000
  553. a=fmtp:101 0-16
  554. a=maxptime:150
  555. a=sendrecv
  556.  
  557. ---
  558. [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
  559. Packet timed out after 32000ms with no response
  560. Really destroying SIP dialog 'kv40PFslKyjCAY7ZiEL9kA..' Method: INVITE
  561. Really destroying SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' Method: REGISTER
  562.  
  563. <--- SIP read from UDP:my_ip:5061 --->
  564.  
  565.  
  566. <------------->
  567.  
  568. <--- SIP read from UDP:my_ip:5061 --->
  569. REGISTER sip:server_ip;transport=UDP SIP/2.0
  570. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8ec7eeb6edc0735a;rport
  571. Max-Forwards: 70
  572. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>
  573. To: <sip:6001@server_ip;transport=UDP>
  574. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  575. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  576. CSeq: 13143 REGISTER
  577. Expires: 60
  578. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  579. User-Agent: Z 5.2.28 rv2.8.114
  580. Authorization: Digest username="6001",realm="asterisk",nonce="67f14424",uri="sip:server_ip;transport=UDP",response="d2df9ffd1eea02262bdaf940de845411",algorithm=MD5
  581. Allow-Events: presence, kpml, talk
  582. Content-Length: 0
  583.  
  584. <------------->
  585. --- (14 headers 0 lines) ---
  586. Sending to my_ip:5061 (NAT)
  587. Sending to my_ip:5061 (NAT)
  588.  
  589. <--- Transmitting (NAT) to my_ip:5061 --->
  590. SIP/2.0 401 Unauthorized
  591. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---8ec7eeb6edc0735a;received=my_ip;rport=5061
  592. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  593. To: <sip:6001@server_ip;transport=UDP>;tag=as1613b627
  594. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  595. CSeq: 13143 REGISTER
  596. Server: Asterisk PBX 16.5.0
  597. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  598. Supported: replaces, timer
  599. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ab2c4b3"
  600. Content-Length: 0
  601.  
  602.  
  603. <------------>
  604. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
  605.  
  606. <--- SIP read from UDP:my_ip:5061 --->
  607. REGISTER sip:server_ip;transport=UDP SIP/2.0
  608. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---dbf8c9aa7345882c;rport
  609. Max-Forwards: 70
  610. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>
  611. To: <sip:6001@server_ip;transport=UDP>
  612. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  613. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  614. CSeq: 13144 REGISTER
  615. Expires: 60
  616. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  617. User-Agent: Z 5.2.28 rv2.8.114
  618. Authorization: Digest username="6001",realm="asterisk",nonce="3ab2c4b3",uri="sip:server_ip;transport=UDP",response="2830ebbdd36787a5ef0e80ef4dd50c96",algorithm=MD5
  619. Allow-Events: presence, kpml, talk
  620. Content-Length: 0
  621.  
  622. <------------->
  623. --- (14 headers 0 lines) ---
  624. Sending to my_ip:5061 (NAT)
  625.  
  626. <--- Transmitting (NAT) to my_ip:5061 --->
  627. SIP/2.0 200 OK
  628. Via: SIP/2.0/UDP client_subnet:5061;branch=z9hG4bK-524287-1---dbf8c9aa7345882c;received=my_ip;rport=5061
  629. From: <sip:6001@server_ip;transport=UDP>;tag=a14d1338
  630. To: <sip:6001@server_ip;transport=UDP>;tag=as1613b627
  631. Call-ID: 8ELdPvZUs4r_Y6LL0D8a_Q..
  632. CSeq: 13144 REGISTER
  633. Server: Asterisk PBX 16.5.0
  634. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  635. Supported: replaces, timer
  636. Expires: 60
  637. Contact: <sip:6001@my_ip:5061;rinstance=019a188f09cfde6b;transport=UDP>;expires=60
  638. Date: Mon, 26 Aug 2019 21:19:24 GMT
  639. Content-Length: 0
  640.  
  641.  
  642. <------------>
  643. Scheduling destruction of SIP dialog '8ELdPvZUs4r_Y6LL0D8a_Q..' in 32000 ms (Method: REGISTER)
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