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  1. [root@localhost ~]# asterisk -rvvvvvvvvv
  2. Asterisk 14.4.0, Copyright (C) 1999 - 2016, Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  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 14.4.0 currently running on localhost (pid = 5277)
  10. [2017-06-04 06:17:15] NOTICE[5344]: res_pjsip/pjsip_distributor.c:536 log_failed_request: Request 'INVITE' from '"1001" <sip:1001@37.61.152.138>' failed for '62.138.14.191:5082' (callid: 58344bf120be560b1137146ca1e94523) - No matching endpoint found
  11. localhost*CLI> sip set debug on
  12. SIP Debugging enabled
  13. localhost*CLI> pjsip set logger on
  14. PJSIP Logging enabled
  15. <--- Received SIP request (894 bytes) from UDP:190.149.80.147:11842 --->
  16. INVITE sip:*60@37.61.152.138 SIP/2.0
  17. Via: SIP/2.0/UDP 192.168.1.3:49849;branch=z9hG4bK-524287-1---6df21f65fd276311;rport
  18. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  19. From: "500"<sip:500@37.61.152.138>;tag=69a4da4f
  20. To: <sip:*60@37.61.152.138>
  21. CSeq: 1 INVITE
  22. Max-Forwards: 70
  23. Contact: <sip:500@192.168.1.3:49849>
  24. Allow: SUBSCRIBE,NOTIFY,INVITE,ACK,CANCEL,BYE,REFER,INFO,OPTIONS,MESSAGE
  25. Supported: replaces
  26. User-Agent: X-Lite release 4.9.8 stamp 84253
  27. Content-Length: 367
  28. Content-Type: application/sdp
  29.  
  30. v=0
  31. o=- 13141030640283850 1 IN IP4 190.149.80.147
  32. s=X-Lite release 4.9.8 stamp 84253
  33. c=IN IP4 190.149.80.147
  34. t=0 0
  35. m=audio 11682 RTP/AVP 9 8 85 120 0 84 3 101
  36. a=rtpmap:85 speex/8000
  37. a=rtpmap:120 opus/48000/2
  38. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  39. a=rtpmap:84 speex/16000
  40. a=rtpmap:101 telephone-event/8000
  41. a=fmtp:101 0-15
  42. a=sendrecv
  43.  
  44. <--- Transmitting SIP response (538 bytes) to UDP:190.149.80.147:11842 --->
  45. SIP/2.0 401 Unauthorized
  46. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---6df21f65fd276311
  47. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  48. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  49. To: <sip:*60@37.61.152.138>;tag=z9hG4bK-524287-1---6df21f65fd276311
  50. CSeq: 1 INVITE
  51. WWW-Authenticate: Digest  realm="asterisk",nonce="1496557051/f2da3b9076ce53c872424062bd7577be",opaque="4b0d7f7d0ec24595",algorithm=md5,qop="auth"
  52. Server: FPBX-13.0.191.10(14.4.0)
  53. Content-Length:  0
  54.  
  55.  
  56. <--- Received SIP request (349 bytes) from UDP:190.149.80.147:11842 --->
  57. ACK sip:*60@37.61.152.138 SIP/2.0
  58. Via: SIP/2.0/UDP 192.168.1.3:49849;branch=z9hG4bK-524287-1---6df21f65fd276311;rport
  59. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  60. From: "500"<sip:500@37.61.152.138>;tag=69a4da4f
  61. To: <sip:*60@37.61.152.138>;tag=z9hG4bK-524287-1---6df21f65fd276311
  62. CSeq: 1 ACK
  63. Max-Forwards: 70
  64. Content-Length: 0
  65.  
  66.  
  67. <--- Received SIP request (1176 bytes) from UDP:190.149.80.147:11842 --->
  68. INVITE sip:*60@37.61.152.138 SIP/2.0
  69. Via: SIP/2.0/UDP 192.168.1.3:49849;branch=z9hG4bK-524287-1---7549a6796d9b967e;rport
  70. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  71. From: "500"<sip:500@37.61.152.138>;tag=69a4da4f
  72. To: <sip:*60@37.61.152.138>
  73. CSeq: 2 INVITE
  74. Max-Forwards: 70
  75. Contact: <sip:500@192.168.1.3:49849>
  76. Allow: SUBSCRIBE,NOTIFY,INVITE,ACK,CANCEL,BYE,REFER,INFO,OPTIONS,MESSAGE
  77. Supported: replaces
  78. User-Agent: X-Lite release 4.9.8 stamp 84253
  79. Authorization: Digest username="500",realm="asterisk",nonce="1496557051/f2da3b9076ce53c872424062bd7577be",uri="sip:*60@37.61.152.138",response="e79abe928c7dbfaf4b4e4d9ee394f657",algorithm=MD5,cnonce="ae27df064bbd7a51486ae59e6690bf5b",opaque="4b0d7f7d0ec24595",qop=auth,nc=00000001
  80. Content-Length: 367
  81. Content-Type: application/sdp
  82.  
  83. v=0
  84. o=- 13141030640283850 1 IN IP4 190.149.80.147
  85. s=X-Lite release 4.9.8 stamp 84253
  86. c=IN IP4 190.149.80.147
  87. t=0 0
  88. m=audio 11682 RTP/AVP 9 8 85 120 0 84 3 101
  89. a=rtpmap:85 speex/8000
  90. a=rtpmap:120 opus/48000/2
  91. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  92. a=rtpmap:84 speex/16000
  93. a=rtpmap:101 telephone-event/8000
  94. a=fmtp:101 0-15
  95. a=sendrecv
  96.  
  97.   == Setting global variable 'SIPDOMAIN' to '37.61.152.138'
  98. <--- Transmitting SIP response (345 bytes) to UDP:190.149.80.147:11842 --->
  99. SIP/2.0 100 Trying
  100. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  101. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  102. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  103. To: <sip:*60@37.61.152.138>
  104. CSeq: 2 INVITE
  105. Server: FPBX-13.0.191.10(14.4.0)
  106. Content-Length:  0
  107.  
  108.  
  109.     -- Executing [*60@from-internal:1] Set("PJSIP/500-00000001", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
  110.     -- Executing [*60@from-internal:2] Set("PJSIP/500-00000001", "CONNECTEDLINE(name,i)=Speaking Clock") in new stack
  111.     -- Executing [*60@from-internal:3] Set("PJSIP/500-00000001", "CONNECTEDLINE(num,i)=*60") in new stack
  112.     -- Executing [*60@from-internal:4] Macro("PJSIP/500-00000001", "user-callerid,") in new stack
  113.     -- Executing [s@macro-user-callerid:1] Set("PJSIP/500-00000001", "TOUCH_MONITOR=1496557051.1") in new stack
  114.     -- Executing [s@macro-user-callerid:2] Set("PJSIP/500-00000001", "AMPUSER=500") in new stack
  115.     -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/500-00000001", "0?report") in new stack
  116.     -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/500-00000001", "1?Set(REALCALLERIDNUM=500)") in new stack
  117.     -- Executing [s@macro-user-callerid:5] Set("PJSIP/500-00000001", "AMPUSER=500") in new stack
  118.     -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/500-00000001", "0?limit") in new stack
  119.     -- Executing [s@macro-user-callerid:7] Set("PJSIP/500-00000001", "AMPUSERCIDNAME=500") in new stack
  120.     -- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/500-00000001", "0?report") in new stack
  121.     -- Executing [s@macro-user-callerid:9] Set("PJSIP/500-00000001", "AMPUSERCID=500") in new stack
  122.     -- Executing [s@macro-user-callerid:10] Set("PJSIP/500-00000001", "__DIAL_OPTIONS=Ttr") in new stack
  123.     -- Executing [s@macro-user-callerid:11] Set("PJSIP/500-00000001", "CALLERID(all)="500" <500>") in new stack
  124.     -- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/500-00000001", "0?limit") in new stack
  125.     -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/500-00000001", "0?Set(GROUP(concurrency_limit)=500)") in new stack
  126.     -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/500-00000001", "0?Set(CHANNEL(language)=)") in new stack
  127.     -- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/500-00000001", "0?continue") in new stack
  128.     -- Executing [s@macro-user-callerid:16] ExecIf("PJSIP/500-00000001", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  129.     -- Executing [s@macro-user-callerid:17] Set("PJSIP/500-00000001", "__TTL=64") in new stack
  130.     -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/500-00000001", "1?continue") in new stack
  131.     -- Goto (macro-user-callerid,s,29)
  132.     -- Executing [s@macro-user-callerid:29] Set("PJSIP/500-00000001", "CALLERID(number)=500") in new stack
  133.     -- Executing [s@macro-user-callerid:30] Set("PJSIP/500-00000001", "CALLERID(name)=500") in new stack
  134.     -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/500-00000001", "0?cnum") in new stack
  135.     -- Executing [s@macro-user-callerid:32] Set("PJSIP/500-00000001", "CDR(cnam)=500") in new stack
  136.     -- Executing [s@macro-user-callerid:33] Set("PJSIP/500-00000001", "CDR(cnum)=500") in new stack
  137.     -- Executing [s@macro-user-callerid:34] Set("PJSIP/500-00000001", "CHANNEL(language)=en") in new stack
  138.     -- Executing [*60@from-internal:5] Answer("PJSIP/500-00000001", "") in new stack
  139. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  140. SIP/2.0 200 OK
  141. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  142. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  143. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  144. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  145. CSeq: 2 INVITE
  146. Server: FPBX-13.0.191.10(14.4.0)
  147. Contact: <sip:37.61.152.13:5060>
  148. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  149. Supported: 100rel, timer, replaces, norefersub
  150. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  151. Content-Type: application/sdp
  152. Content-Length:   277
  153.  
  154. v=0
  155. o=- 2672554186 3 IN IP4 192.168.1.150
  156. s=Asterisk
  157. c=IN IP4 192.168.1.150
  158. t=0 0
  159. m=audio 15328 RTP/AVP 0 8 3 101
  160. a=rtpmap:0 PCMU/8000
  161. a=rtpmap:8 PCMA/8000
  162. a=rtpmap:3 GSM/8000
  163. a=rtpmap:101 telephone-event/8000
  164. a=fmtp:101 0-16
  165. a=ptime:20
  166. a=maxptime:150
  167. a=sendrecv
  168.  
  169. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  170. SIP/2.0 200 OK
  171. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  172. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  173. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  174. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  175. CSeq: 2 INVITE
  176. Server: FPBX-13.0.191.10(14.4.0)
  177. Contact: <sip:37.61.152.13:5060>
  178. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  179. Supported: 100rel, timer, replaces, norefersub
  180. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  181. Content-Type: application/sdp
  182. Content-Length:   277
  183.  
  184. v=0
  185. o=- 2672554186 3 IN IP4 192.168.1.150
  186. s=Asterisk
  187. c=IN IP4 192.168.1.150
  188. t=0 0
  189. m=audio 15328 RTP/AVP 0 8 3 101
  190. a=rtpmap:0 PCMU/8000
  191. a=rtpmap:8 PCMA/8000
  192. a=rtpmap:3 GSM/8000
  193. a=rtpmap:101 telephone-event/8000
  194. a=fmtp:101 0-16
  195. a=ptime:20
  196. a=maxptime:150
  197. a=sendrecv
  198.  
  199.     -- Executing [*60@from-internal:6] Wait("PJSIP/500-00000001", "1") in new stack
  200. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  201. SIP/2.0 200 OK
  202. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  203. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  204. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  205. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  206. CSeq: 2 INVITE
  207. Server: FPBX-13.0.191.10(14.4.0)
  208. Contact: <sip:37.61.152.13:5060>
  209. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  210. Supported: 100rel, timer, replaces, norefersub
  211. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  212. Content-Type: application/sdp
  213. Content-Length:   277
  214.  
  215. v=0
  216. o=- 2672554186 3 IN IP4 192.168.1.150
  217. s=Asterisk
  218. c=IN IP4 192.168.1.150
  219. t=0 0
  220. m=audio 15328 RTP/AVP 0 8 3 101
  221. a=rtpmap:0 PCMU/8000
  222. a=rtpmap:8 PCMA/8000
  223. a=rtpmap:3 GSM/8000
  224. a=rtpmap:101 telephone-event/8000
  225. a=fmtp:101 0-16
  226. a=ptime:20
  227. a=maxptime:150
  228. a=sendrecv
  229.  
  230.     -- Executing [*60@from-internal:7] Set("PJSIP/500-00000001", "NumLoops=0") in new stack
  231.     -- Executing [*60@from-internal:8] Set("PJSIP/500-00000001", "FutureTime=1496557060") in new stack
  232.     -- Executing [*60@from-internal:9] Set("PJSIP/500-00000001", "FutureTimeMod=0") in new stack
  233.     -- Executing [*60@from-internal:10] Set("PJSIP/500-00000001", "FutureTime=1496557070") in new stack
  234.     -- Executing [*60@from-internal:11] Gosub("PJSIP/500-00000001", "sub-hr12format,s,1()") in new stack
  235.     -- Executing [s@sub-hr12format:1] GotoIf("PJSIP/500-00000001", "1?sub-hr12format,en,1:sub-hr12format,en,1") in new stack
  236.     -- Goto (sub-hr12format,en,1)
  237.     -- Executing [en@sub-hr12format:1] Playback("PJSIP/500-00000001", "at-tone-time-exactly") in new stack
  238.     -- <PJSIP/500-00000001> Playing 'at-tone-time-exactly.ulaw' (language 'en')
  239. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  240. SIP/2.0 200 OK
  241. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  242. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  243. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  244. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  245. CSeq: 2 INVITE
  246. Server: FPBX-13.0.191.10(14.4.0)
  247. Contact: <sip:37.61.152.13:5060>
  248. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  249. Supported: 100rel, timer, replaces, norefersub
  250. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  251. Content-Type: application/sdp
  252. Content-Length:   277
  253.  
  254. v=0
  255. o=- 2672554186 3 IN IP4 192.168.1.150
  256. s=Asterisk
  257. c=IN IP4 192.168.1.150
  258. t=0 0
  259. m=audio 15328 RTP/AVP 0 8 3 101
  260. a=rtpmap:0 PCMU/8000
  261. a=rtpmap:8 PCMA/8000
  262. a=rtpmap:3 GSM/8000
  263. a=rtpmap:101 telephone-event/8000
  264. a=fmtp:101 0-16
  265. a=ptime:20
  266. a=maxptime:150
  267. a=sendrecv
  268.  
  269.     -- Executing [en@sub-hr12format:2] SayUnixTime("PJSIP/500-00000001", "1496557070,,IM 'vm-and' S 'seconds' p") in new stack
  270.     -- <PJSIP/500-00000001> Playing 'digits/6.ulaw' (language 'en')
  271.     -- <PJSIP/500-00000001> Playing 'digits/17.ulaw' (language 'en')
  272.     -- <PJSIP/500-00000001> Playing 'vm-and.ulaw' (language 'en')
  273. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  274. SIP/2.0 200 OK
  275. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  276. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  277. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  278. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  279. CSeq: 2 INVITE
  280. Server: FPBX-13.0.191.10(14.4.0)
  281. Contact: <sip:37.61.152.13:5060>
  282. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  283. Supported: 100rel, timer, replaces, norefersub
  284. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  285. Content-Type: application/sdp
  286. Content-Length:   277
  287.  
  288. v=0
  289. o=- 2672554186 3 IN IP4 192.168.1.150
  290. s=Asterisk
  291. c=IN IP4 192.168.1.150
  292. t=0 0
  293. m=audio 15328 RTP/AVP 0 8 3 101
  294. a=rtpmap:0 PCMU/8000
  295. a=rtpmap:8 PCMA/8000
  296. a=rtpmap:3 GSM/8000
  297. a=rtpmap:101 telephone-event/8000
  298. a=fmtp:101 0-16
  299. a=ptime:20
  300. a=maxptime:150
  301. a=sendrecv
  302.  
  303.     -- <PJSIP/500-00000001> Playing 'digits/50.ulaw' (language 'en')
  304.     -- <PJSIP/500-00000001> Playing 'seconds.ulaw' (language 'en')
  305.     -- <PJSIP/500-00000001> Playing 'digits/a-m.ulaw' (language 'en')
  306.     -- Executing [en@sub-hr12format:3] Return("PJSIP/500-00000001", "") in new stack
  307.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=8") in new stack
  308.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  309.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  310. <--- Received SIP request (765 bytes) from UDP:62.138.14.191:5074 --->
  311. INVITE sip:2222#0048483888460@37.61.152.138 SIP/2.0
  312. To: 2222#0048483888460<sip:2222#0048483888460@37.61.152.138>
  313. From: 1001<sip:1001@37.61.152.138>;tag=abfe8383
  314. Via: SIP/2.0/UDP 62.138.14.191:5074;branch=z9hG4bK-1b4bb0f1820da783a3a6e4aae4f85d23;rport
  315. Call-ID: 1b4bb0f1820da783a3a6e4aae4f85d23
  316. CSeq: 1 INVITE
  317. Contact: <sip:1001@62.138.14.191:5074>
  318. Max-Forwards: 70
  319. Allow: INVITE, ACK, CANCEL, BYE
  320. User-Agent: sipcli/v1.8
  321. Content-Type: application/sdp
  322. Content-Length: 281
  323.  
  324. v=0
  325. o=sipcli-Session 23957935 1323117518 IN IP4 62.138.14.191
  326. s=sipcli
  327. c=IN IP4 62.138.14.191
  328. t=0 0
  329. m=audio 5075 RTP/AVP 18 0 8 101
  330. a=fmtp:101 0-15
  331. a=rtpmap:18 G729/8000
  332. a=rtpmap:0 PCMU/8000
  333. a=rtpmap:8 PCMA/8000
  334. a=rtpmap:101 telephone-event/8000
  335. a=ptime:20
  336. a=sendrecv
  337.  
  338. [2017-06-04 06:17:42] NOTICE[5344]: res_pjsip/pjsip_distributor.c:536 log_failed_request: Request 'INVITE' from '"1001" <sip:1001@37.61.152.138>' failed for '62.138.14.191:5074' (callid: 1b4bb0f1820da783a3a6e4aae4f85d23) - No matching endpoint found
  339. <--- Transmitting SIP response (571 bytes) to UDP:62.138.14.191:5074 --->
  340. SIP/2.0 401 Unauthorized
  341. Via: SIP/2.0/UDP 62.138.14.191:5074;rport=5074;received=62.138.14.191;branch=z9hG4bK-1b4bb0f1820da783a3a6e4aae4f85d23
  342. Call-ID: 1b4bb0f1820da783a3a6e4aae4f85d23
  343. From: "1001" <sip:1001@37.61.152.138>;tag=abfe8383
  344. To: "2222#0048483888460" <sip:2222%230048483888460@37.61.152.138>;tag=z9hG4bK-1b4bb0f1820da783a3a6e4aae4f85d23
  345. CSeq: 1 INVITE
  346. WWW-Authenticate: Digest  realm="asterisk",nonce="1496557062/f36c01640b75f70fa67d3af8513ae11e",opaque="0a70da6b7484cbda",algorithm=md5,qop="auth"
  347. Server: FPBX-13.0.191.10(14.4.0)
  348. Content-Length:  0
  349.  
  350.  
  351. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  352. SIP/2.0 200 OK
  353. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  354. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  355. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  356. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  357. CSeq: 2 INVITE
  358. Server: FPBX-13.0.191.10(14.4.0)
  359. Contact: <sip:37.61.152.13:5060>
  360. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  361. Supported: 100rel, timer, replaces, norefersub
  362. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  363. Content-Type: application/sdp
  364. Content-Length:   277
  365.  
  366. v=0
  367. o=- 2672554186 3 IN IP4 192.168.1.150
  368. s=Asterisk
  369. c=IN IP4 192.168.1.150
  370. t=0 0
  371. m=audio 15328 RTP/AVP 0 8 3 101
  372. a=rtpmap:0 PCMU/8000
  373. a=rtpmap:8 PCMA/8000
  374. a=rtpmap:3 GSM/8000
  375. a=rtpmap:101 telephone-event/8000
  376. a=fmtp:101 0-16
  377. a=ptime:20
  378. a=maxptime:150
  379. a=sendrecv
  380.  
  381.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  382.     -- Goto (from-internal,*60,12)
  383.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=7") in new stack
  384.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  385.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  386.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  387.     -- Goto (from-internal,*60,12)
  388.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=6") in new stack
  389.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  390.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  391.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  392.     -- Goto (from-internal,*60,12)
  393.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=5") in new stack
  394.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  395.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  396.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  397.     -- Goto (from-internal,*60,12)
  398.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=4") in new stack
  399.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  400.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  401. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  402. SIP/2.0 200 OK
  403. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  404. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  405. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  406. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  407. CSeq: 2 INVITE
  408. Server: FPBX-13.0.191.10(14.4.0)
  409. Contact: <sip:37.61.152.13:5060>
  410. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  411. Supported: 100rel, timer, replaces, norefersub
  412. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  413. Content-Type: application/sdp
  414. Content-Length:   277
  415.  
  416. v=0
  417. o=- 2672554186 3 IN IP4 192.168.1.150
  418. s=Asterisk
  419. c=IN IP4 192.168.1.150
  420. t=0 0
  421. m=audio 15328 RTP/AVP 0 8 3 101
  422. a=rtpmap:0 PCMU/8000
  423. a=rtpmap:8 PCMA/8000
  424. a=rtpmap:3 GSM/8000
  425. a=rtpmap:101 telephone-event/8000
  426. a=fmtp:101 0-16
  427. a=ptime:20
  428. a=maxptime:150
  429. a=sendrecv
  430.  
  431.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  432.     -- Goto (from-internal,*60,12)
  433.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=3") in new stack
  434.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  435.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  436.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  437.     -- Goto (from-internal,*60,12)
  438.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=2") in new stack
  439.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  440.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  441.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  442.     -- Goto (from-internal,*60,12)
  443.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=1") in new stack
  444.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "0?playbeep") in new stack
  445.     -- Executing [*60@from-internal:14] Wait("PJSIP/500-00000001", "1") in new stack
  446.     -- Executing [*60@from-internal:15] Goto("PJSIP/500-00000001", "waitloop") in new stack
  447.     -- Goto (from-internal,*60,12)
  448.     -- Executing [*60@from-internal:12] Set("PJSIP/500-00000001", "TimeLeft=0") in new stack
  449.     -- Executing [*60@from-internal:13] GotoIf("PJSIP/500-00000001", "1?playbeep") in new stack
  450.     -- Goto (from-internal,*60,16)
  451.     -- Executing [*60@from-internal:16] Playback("PJSIP/500-00000001", "beep") in new stack
  452.     -- <PJSIP/500-00000001> Playing 'beep.ulaw' (language 'en')
  453. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  454. SIP/2.0 200 OK
  455. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  456. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  457. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  458. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  459. CSeq: 2 INVITE
  460. Server: FPBX-13.0.191.10(14.4.0)
  461. Contact: <sip:37.61.152.13:5060>
  462. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  463. Supported: 100rel, timer, replaces, norefersub
  464. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  465. Content-Type: application/sdp
  466. Content-Length:   277
  467.  
  468. v=0
  469. o=- 2672554186 3 IN IP4 192.168.1.150
  470. s=Asterisk
  471. c=IN IP4 192.168.1.150
  472. t=0 0
  473. m=audio 15328 RTP/AVP 0 8 3 101
  474. a=rtpmap:0 PCMU/8000
  475. a=rtpmap:8 PCMA/8000
  476. a=rtpmap:3 GSM/8000
  477. a=rtpmap:101 telephone-event/8000
  478. a=fmtp:101 0-16
  479. a=ptime:20
  480. a=maxptime:150
  481. a=sendrecv
  482.  
  483.     -- Executing [*60@from-internal:17] Wait("PJSIP/500-00000001", "5") in new stack
  484. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  485. SIP/2.0 200 OK
  486. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  487. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  488. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  489. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  490. CSeq: 2 INVITE
  491. Server: FPBX-13.0.191.10(14.4.0)
  492. Contact: <sip:37.61.152.13:5060>
  493. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  494. Supported: 100rel, timer, replaces, norefersub
  495. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  496. Content-Type: application/sdp
  497. Content-Length:   277
  498.  
  499. v=0
  500. o=- 2672554186 3 IN IP4 192.168.1.150
  501. s=Asterisk
  502. c=IN IP4 192.168.1.150
  503. t=0 0
  504. m=audio 15328 RTP/AVP 0 8 3 101
  505. a=rtpmap:0 PCMU/8000
  506. a=rtpmap:8 PCMA/8000
  507. a=rtpmap:3 GSM/8000
  508. a=rtpmap:101 telephone-event/8000
  509. a=fmtp:101 0-16
  510. a=ptime:20
  511. a=maxptime:150
  512. a=sendrecv
  513.  
  514.     -- Executing [*60@from-internal:18] Set("PJSIP/500-00000001", "NumLoops=1") in new stack
  515.     -- Executing [*60@from-internal:19] GotoIf("PJSIP/500-00000001", "1?start") in new stack
  516.     -- Goto (from-internal,*60,8)
  517.     -- Executing [*60@from-internal:8] Set("PJSIP/500-00000001", "FutureTime=1496557084") in new stack
  518.     -- Executing [*60@from-internal:9] Set("PJSIP/500-00000001", "FutureTimeMod=4") in new stack
  519.     -- Executing [*60@from-internal:10] Set("PJSIP/500-00000001", "FutureTime=1496557090") in new stack
  520.     -- Executing [*60@from-internal:11] Gosub("PJSIP/500-00000001", "sub-hr12format,s,1()") in new stack
  521.     -- Executing [s@sub-hr12format:1] GotoIf("PJSIP/500-00000001", "1?sub-hr12format,en,1:sub-hr12format,en,1") in new stack
  522.     -- Goto (sub-hr12format,en,1)
  523.     -- Executing [en@sub-hr12format:1] Playback("PJSIP/500-00000001", "at-tone-time-exactly") in new stack
  524.     -- <PJSIP/500-00000001> Playing 'at-tone-time-exactly.ulaw' (language 'en')
  525. <--- Transmitting SIP response (949 bytes) to UDP:190.149.80.147:11842 --->
  526. SIP/2.0 200 OK
  527. Via: SIP/2.0/UDP 192.168.1.3:49849;rport=11842;received=190.149.80.147;branch=z9hG4bK-524287-1---7549a6796d9b967e
  528. Call-ID: 84253ZjY3YzJiYjA5ODYxZmUzNWQ0NDQwOTkwN2U5ZTVjNDA
  529. From: "500" <sip:500@37.61.152.138>;tag=69a4da4f
  530. To: <sip:*60@37.61.152.138>;tag=623bce2f-2cd0-44e4-ad0e-7bfbae7199dd
  531. CSeq: 2 INVITE
  532. Server: FPBX-13.0.191.10(14.4.0)
  533. Contact: <sip:37.61.152.13:5060>
  534. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  535. Supported: 100rel, timer, replaces, norefersub
  536. P-Asserted-Identity: "Speaking Clock" <sip:*60@37.61.152.138>
  537. Content-Type: application/sdp
  538. Content-Length:   277
  539.  
  540. v=0
  541. o=- 2672554186 3 IN IP4 192.168.1.150
  542. s=Asterisk
  543. c=IN IP4 192.168.1.150
  544. t=0 0
  545. m=audio 15328 RTP/AVP 0 8 3 101
  546. a=rtpmap:0 PCMU/8000
  547. a=rtpmap:8 PCMA/8000
  548. a=rtpmap:3 GSM/8000
  549. a=rtpmap:101 telephone-event/8000
  550. a=fmtp:101 0-16
  551. a=ptime:20
  552. a=maxptime:150
  553. a=sendrecv
  554.  
  555. localhost*CLI> exit
  556. Asterisk cleanly ending (0).
  557. Executing last minute cleanups

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