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zejeske
Apr 30th, 2018
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  1. [root@localhost ~]# asterisk -r
  2. Asterisk 11.25.3, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  3. Created by Mark Spencer <[email protected]>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 11.25.3 currently running on localhost (pid = 2499)
  10. localhost*CLI> SIP SET DEBUG peer tk_saida
  11. SIP Debugging Enabled for IP: 200.184.77.169
  12. Audio is at 16702
  13. Adding codec 100004 (alaw) to SDP
  14. Adding codec 100003 (ulaw) to SDP
  15. Adding codec 100008 (g729) to SDP
  16. Adding non-codec 0x1 (telephone-event) to SDP
  17. Reliably Transmitting (NAT) to 200.184.77.169:5060:
  18. INVITE sip:[email protected] SIP/2.0
  19. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;rport
  20. Max-Forwards: 70
  21. From: <sip:[email protected]>;tag=as20b1d26c
  22. To: <sip:[email protected]>
  23. Contact: <sip:[email protected]:5060>
  24. Call-ID: [email protected]:5060
  25. CSeq: 102 INVITE
  26. User-Agent: FPBX-2.11.0(11.25.3)
  27. Date: Mon, 30 Apr 2018 14:42:55 GMT
  28. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  29. Supported: replaces, timer
  30. Content-Type: application/sdp
  31. Content-Length: 309
  32.  
  33. v=0
  34. o=root 1331459369 1331459369 IN IP4 192.168.1.111
  35. s=Asterisk PBX 11.25.3
  36. c=IN IP4 192.168.1.111
  37. t=0 0
  38. m=audio 16702 RTP/AVP 8 0 18 101
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:0 PCMU/8000
  41. a=rtpmap:18 G729/8000
  42. a=fmtp:18 annexb=no
  43. a=rtpmap:101 telephone-event/8000
  44. a=fmtp:101 0-16
  45. a=ptime:20
  46. a=sendrecv
  47.  
  48. ---
  49.  
  50. <--- SIP read from UDP:200.184.77.169:5060 --->
  51. SIP/2.0 407 Proxy Authentication Required
  52. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;received=187.85.174.1 34;rport=5060
  53. From: <sip:[email protected]>;tag=as20b1d26c
  54. To: <sip:[email protected]>;tag=as6112cc95
  55. Call-ID: [email protected]:5060
  56. CSeq: 102 INVITE
  57. User-Agent: InforSOLutions Billing
  58. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  59. Supported: replaces
  60. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f043f66"
  61. Content-Length: 0
  62.  
  63. <------------->
  64. --- (11 headers 0 lines) ---
  65. Transmitting (NAT) to 200.184.77.169:5060:
  66. ACK sip:[email protected] SIP/2.0
  67. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;rport
  68. Max-Forwards: 70
  69. From: <sip:[email protected]>;tag=as20b1d26c
  70. To: <sip:[email protected]>;tag=as6112cc95
  71. Contact: <sip:[email protected]:5060>
  72. Call-ID: [email protected]:5060
  73. CSeq: 102 ACK
  74. User-Agent: FPBX-2.11.0(11.25.3)
  75. Content-Length: 0
  76.  
  77.  
  78. ---
  79. Audio is at 16702
  80. Adding codec 100004 (alaw) to SDP
  81. Adding codec 100003 (ulaw) to SDP
  82. Adding codec 100008 (g729) to SDP
  83. Adding non-codec 0x1 (telephone-event) to SDP
  84. Reliably Transmitting (NAT) to 200.184.77.169:5060:
  85. INVITE sip:[email protected] SIP/2.0
  86. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
  87. Max-Forwards: 70
  88. From: <sip:[email protected]>;tag=as20b1d26c
  89. To: <sip:[email protected]>
  90. Contact: <sip:[email protected]:5060>
  91. Call-ID: [email protected]:5060
  92. CSeq: 103 INVITE
  93. User-Agent: FPBX-2.11.0(11.25.3)
  94. Proxy-Authorization: Digest username="5076", realm="asterisk", algorithm=MD5, ur i="sip:[email protected]", nonce="2f043f66", response="f5299ae6e1cba9068ff 8d67f63b00add"
  95. Date: Mon, 30 Apr 2018 14:42:55 GMT
  96. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  97. Supported: replaces, timer
  98. Content-Type: application/sdp
  99. Content-Length: 309
  100.  
  101. v=0
  102. o=root 1331459369 1331459370 IN IP4 192.168.1.111
  103. s=Asterisk PBX 11.25.3
  104. c=IN IP4 192.168.1.111
  105. t=0 0
  106. m=audio 16702 RTP/AVP 8 0 18 101
  107. a=rtpmap:8 PCMA/8000
  108. a=rtpmap:0 PCMU/8000
  109. a=rtpmap:18 G729/8000
  110. a=fmtp:18 annexb=no
  111. a=rtpmap:101 telephone-event/8000
  112. a=fmtp:101 0-16
  113. a=ptime:20
  114. a=sendrecv
  115.  
  116. ---
  117.  
  118. <--- SIP read from UDP:200.184.77.169:5060 --->
  119. SIP/2.0 100 Trying
  120. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
  121. From: <sip:[email protected]>;tag=as20b1d26c
  122. To: <sip:[email protected]>
  123. Call-ID: [email protected]:5060
  124. CSeq: 103 INVITE
  125. User-Agent: InforSOLutions Billing
  126. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  127. Supported: replaces
  128. Contact: <sip:[email protected]>
  129. Content-Length: 0
  130.  
  131. <------------->
  132. --- (11 headers 0 lines) ---
  133. [2018-04-30 11:42:59] WARNING[6332][C-00000040]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  134.  
  135. <--- SIP read from UDP:200.184.77.169:5060 --->
  136. SIP/2.0 180 Ringing
  137. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
  138. From: <sip:[email protected]>;tag=as20b1d26c
  139. To: <sip:[email protected]>;tag=as4b2a9893
  140. Call-ID: [email protected]:5060
  141. CSeq: 103 INVITE
  142. User-Agent: InforSOLutions Billing
  143. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  144. Supported: replaces
  145. Contact: <sip:[email protected]>
  146. Content-Length: 0
  147.  
  148. <------------->
  149. --- (11 headers 0 lines) ---
  150. list_route: hop: <sip:[email protected]>
  151.  
  152. <--- SIP read from UDP:200.184.77.169:5060 --->
  153. SIP/2.0 183 Session Progress
  154. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
  155. From: <sip:[email protected]>;tag=as20b1d26c
  156. To: <sip:[email protected]>;tag=as4b2a9893
  157. Call-ID: [email protected]:5060
  158. CSeq: 103 INVITE
  159. User-Agent: InforSOLutions Billing
  160. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  161. Supported: replaces
  162. Contact: <sip:[email protected]>
  163. Content-Type: application/sdp
  164. Content-Length: 313
  165.  
  166. v=0
  167. o=root 3360 3360 IN IP4 200.184.77.169
  168. s=session
  169. c=IN IP4 200.184.77.169
  170. t=0 0
  171. m=audio 16820 RTP/AVP 18 8 0 101
  172. a=rtpmap:18 G729/8000
  173. a=fmtp:18 annexb=no
  174. a=rtpmap:8 PCMA/8000
  175. a=rtpmap:0 PCMU/8000
  176. a=rtpmap:101 telephone-event/8000
  177. a=fmtp:101 0-16
  178. a=silenceSupp:off - - - -
  179. a=ptime:20
  180. a=sendrecv
  181.  
  182. <------------->
  183. --- (12 headers 15 lines) ---
  184. list_route: hop: <sip:[email protected]>
  185. Found RTP audio format 18
  186. Found RTP audio format 8
  187. Found RTP audio format 0
  188. Found RTP audio format 101
  189. Found audio description format G729 for ID 18
  190. Found audio description format PCMA for ID 8
  191. Found audio description format PCMU for ID 0
  192. Found audio description format telephone-event for ID 101
  193. Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothin g)/text=(nothing), combined - (ulaw|alaw|g729)
  194. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
  195. Peer audio RTP is at port 200.184.77.169:16820
  196. Scheduling destruction of SIP dialog '[email protected] .111:5060' in 6400 ms (Method: INVITE)
  197. Reliably Transmitting (NAT) to 200.184.77.169:5060:
  198. CANCEL sip:[email protected] SIP/2.0
  199. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
  200. Max-Forwards: 70
  201. From: <sip:[email protected]>;tag=as20b1d26c
  202. To: <sip:[email protected]>
  203. Call-ID: [email protected]:5060
  204. CSeq: 103 CANCEL
  205. User-Agent: FPBX-2.11.0(11.25.3)
  206. Content-Length: 0
  207.  
  208.  
  209. ---
  210. Scheduling destruction of SIP dialog '[email protected] .111:5060' in 6400 ms (Method: INVITE)
  211.  
  212. <--- SIP read from UDP:200.184.77.169:5060 --->
  213. SIP/2.0 487 Request Terminated
  214. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
  215. From: <sip:[email protected]>;tag=as20b1d26c
  216. To: <sip:[email protected]>;tag=as4b2a9893
  217. Call-ID: [email protected]:5060
  218. CSeq: 103 INVITE
  219. User-Agent: InforSOLutions Billing
  220. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  221. Supported: replaces
  222. Content-Length: 0
  223.  
  224. <------------->
  225. --- (10 headers 0 lines) ---
  226. Transmitting (NAT) to 200.184.77.169:5060:
  227. ACK sip:[email protected] SIP/2.0
  228. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
  229. Max-Forwards: 70
  230. From: <sip:[email protected]>;tag=as20b1d26c
  231. To: <sip:[email protected]>;tag=as4b2a9893
  232. Contact: <sip:[email protected]:5060>
  233. Call-ID: [email protected]:5060
  234. CSeq: 103 ACK
  235. User-Agent: FPBX-2.11.0(11.25.3)
  236. Content-Length: 0
  237.  
  238.  
  239. ---
  240. Scheduling destruction of SIP dialog '[email protected] .111:5060' in 6400 ms (Method: INVITE)
  241. Retransmitting #1 (NAT) to 200.184.77.169:5060:
  242. CANCEL sip:[email protected] SIP/2.0
  243. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
  244. Max-Forwards: 70
  245. From: <sip:[email protected]>;tag=as20b1d26c
  246. To: <sip:[email protected]>
  247. Call-ID: [email protected]:5060
  248. CSeq: 103 CANCEL
  249. User-Agent: FPBX-2.11.0(11.25.3)
  250. Content-Length: 0
  251.  
  252.  
  253. ---
  254.  
  255. <--- SIP read from UDP:200.184.77.169:5060 --->
  256. SIP/2.0 481 Call leg/transaction does not exist
  257. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
  258. From: <sip:[email protected]>;tag=as20b1d26c
  259. To: <sip:[email protected]>;tag=as709ee3d5
  260. Call-ID: [email protected]:5060
  261. CSeq: 103 CANCEL
  262. User-Agent: InforSOLutions Billing
  263. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  264. Supported: replaces
  265. Content-Length: 0
  266.  
  267. <------------->
  268. --- (10 headers 0 lines) ---
  269. [2018-04-30 11:43:03] WARNING[2586][C-0000003f]: chan_sip.c:24287 handle_respons e: Remote host can't match request CANCEL to call '660c74301af7b2086d94e94261191 [email protected]:5060'. Giving up.
  270. Really destroying SIP dialog '[email protected]' M ethod: OPTIONS
  271. Really destroying SIP dialog '[email protected]:506 0' Method: INVITE
  272. Audio is at 12698
  273. Adding codec 100004 (alaw) to SDP
  274. Adding codec 100003 (ulaw) to SDP
  275. Adding codec 100008 (g729) to SDP
  276. Adding non-codec 0x1 (telephone-event) to SDP
  277. Reliably Transmitting (NAT) to 200.184.77.169:5060:
  278. INVITE sip:[email protected] SIP/2.0
  279. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;rport
  280. Max-Forwards: 70
  281. From: <sip:[email protected]>;tag=as2636fbad
  282. To: <sip:[email protected]>
  283. Contact: <sip:[email protected]:5060>
  284. Call-ID: [email protected]:5060
  285. CSeq: 102 INVITE
  286. User-Agent: FPBX-2.11.0(11.25.3)
  287. Date: Mon, 30 Apr 2018 14:43:13 GMT
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  289. Supported: replaces, timer
  290. Content-Type: application/sdp
  291. Content-Length: 307
  292.  
  293. v=0
  294. o=root 878878280 878878280 IN IP4 192.168.1.111
  295. s=Asterisk PBX 11.25.3
  296. c=IN IP4 192.168.1.111
  297. t=0 0
  298. m=audio 12698 RTP/AVP 8 0 18 101
  299. a=rtpmap:8 PCMA/8000
  300. a=rtpmap:0 PCMU/8000
  301. a=rtpmap:18 G729/8000
  302. a=fmtp:18 annexb=no
  303. a=rtpmap:101 telephone-event/8000
  304. a=fmtp:101 0-16
  305. a=ptime:20
  306. a=sendrecv
  307.  
  308. ---
  309.  
  310. <--- SIP read from UDP:200.184.77.169:5060 --->
  311. SIP/2.0 100 Trying
  312. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;received=187.85.174.1 34;rport=5060
  313. From: <sip:[email protected]>;tag=as2636fbad
  314. To: <sip:[email protected]>
  315. Call-ID: [email protected]:5060
  316. CSeq: 102 INVITE
  317. User-Agent: InforSOLutions Billing
  318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  319. Supported: replaces
  320. Contact: <sip:[email protected]>
  321. Content-Length: 0
  322.  
  323. <------------->
  324. --- (11 headers 0 lines) ---
  325.  
  326. <--- SIP read from UDP:200.184.77.169:5060 --->
  327. SIP/2.0 603 Declined
  328. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;received=187.85.174.1 34;rport=5060
  329. From: <sip:[email protected]>;tag=as2636fbad
  330. To: <sip:[email protected]>;tag=as4827d7e2
  331. Call-ID: [email protected]:5060
  332. CSeq: 102 INVITE
  333. User-Agent: InforSOLutions Billing
  334. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  335. Supported: replaces
  336. Content-Length: 0
  337.  
  338. <------------->
  339. --- (10 headers 0 lines) ---
  340. Transmitting (NAT) to 200.184.77.169:5060:
  341. ACK sip:[email protected] SIP/2.0
  342. Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;rport
  343. Max-Forwards: 70
  344. From: <sip:[email protected]>;tag=as2636fbad
  345. To: <sip:[email protected]>;tag=as4827d7e2
  346. Contact: <sip:[email protected]:5060>
  347. Call-ID: [email protected]:5060
  348. CSeq: 102 ACK
  349. User-Agent: FPBX-2.11.0(11.25.3)
  350. Content-Length: 0
  351.  
  352.  
  353. ---
  354. Really destroying SIP dialog '[email protected]:506 0' Method: INVITE
  355. [2018-04-30 11:43:18] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"204" <sip:[email protected]>;tag=2789253 13'
  356. [2018-04-30 11:43:18] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"47 99985 9396"<sip:[email protected]:506 0;transport=UDP>;tag=88594b00'
  357. [2018-04-30 11:43:19] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"47 99985 9396"<sip:[email protected]:506 0;transport=UDP>;tag=88594b00'
  358. localhost*CLI>
  359. Disconnected from Asterisk server
  360. Asterisk cleanly ending (0).
  361. Executing last minute cleanups
  362. [root@localhost ~]#
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