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  1. =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.08.26 22:28:18 =~=~=~=~=~=~=~=~=~=~=~=
  2. login as: root
  3. root@pbx.hecint.com's password:
  4. Last login: Fri Aug 26 16:51:05 2011 from aldur.hecint.com
  5.  
  6. ]0;root@pbx:~[root@pbx ~]# asterisk -rvvvvvvv
  7. Asterisk 1.6.2.15, Copyright (C) 1999 - 2010 Digium, Inc. and others.
  8. Created by Mark Spencer <markster@digium.com>
  9. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  10. This is free software, with components licensed under the GNU General Public
  11. License version 2 and other licenses; you are welcome to redistribute it under
  12. certain conditions. Type 'core show license' for details.
  13. =========================================================================
  14. == Parsing '/etc/asterisk/asterisk.conf': == Found
  15. Connected to Asterisk 1.6.2.15 currently running on pbx (pid = 20793)
  16. pbx*CLI>
  17. Verbosity is at least 7
  18.  
  19. pbx*CLI>
  20.  -- Remote UNIX connection
  21.  
  22. pbx*CLI> clear now
  23. REGISTER 11 headers, 0 lines
  24.  
  25. pbx*CLI> clear now
  26. Reliably Transmitting (NAT) to 216.58.0.51:5060:
  27. REGISTER sip:216.58.0.51 SIP/2.0
  28.  
  29. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK77a5bdaf;rport
  30.  
  31. Max-Forwards: 70
  32.  
  33. From: <sip:6477232824@216.58.0.51>;tag=as596096c9
  34.  
  35. To: <sip:6477232824@216.58.0.51>
  36.  
  37. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  38.  
  39. CSeq: 116 REGISTER
  40.  
  41. User-Agent: FPBX-2.9.0(1.6.2.15)
  42.  
  43. Authorization: Digest username="6477232824", realm="cia.com", algorithm=MD5, uri="sip:216.58.0.51", nonce="4c686fd0", response="4403b1d97ce0acc44a11d24bec7160f5"
  44.  
  45. Expires: 120
  46.  
  47. Contact: <sip:6477232824@99.227.42.4>
  48.  
  49. Content-Length: 0
  50.  
  51.  
  52.  
  53.  
  54. ---
  55.  
  56. pbx*CLI> clear now
  57. 
  58. <--- SIP read from UDP:216.58.0.51:5060 --->
  59. SIP/2.0 100 Trying
  60.  
  61. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK77a5bdaf;received=99.227.42.4;rport=5060
  62.  
  63. From: <sip:6477232824@216.58.0.51>;tag=as596096c9
  64.  
  65. To: <sip:6477232824@216.58.0.51>
  66.  
  67. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  68.  
  69. CSeq: 116 REGISTER
  70.  
  71. User-Agent: CIA.com PBX
  72.  
  73. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  74.  
  75. Supported: replaces
  76.  
  77. Contact: <sip:6477232824@216.58.0.51>
  78.  
  79. Content-Length: 0
  80.  
  81.  
  82.  
  83.  
  84. <------------->
  85.  
  86. pbx*CLI> clear now
  87. --- (11 headers 0 lines) ---
  88.  
  89. pbx*CLI> clear now
  90. 
  91. <--- SIP read from UDP:216.58.0.51:5060 --->
  92. SIP/2.0 401 Unauthorized
  93.  
  94. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK77a5bdaf;received=99.227.42.4;rport=5060
  95.  
  96. From: <sip:6477232824@216.58.0.51>;tag=as596096c9
  97.  
  98. To: <sip:6477232824@216.58.0.51>;tag=as76e4c03c
  99.  
  100. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  101.  
  102. CSeq: 116 REGISTER
  103.  
  104. User-Agent: CIA.com PBX
  105.  
  106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  107.  
  108. Supported: replaces
  109.  
  110. WWW-Authenticate: Digest algorithm=MD5, realm="cia.com", nonce="27ba261c"
  111.  
  112. Content-Length: 0
  113.  
  114.  
  115.  
  116.  
  117. <------------->
  118.  
  119. pbx*CLI> clear now
  120. --- (11 headers 0 lines) ---
  121. Responding to challenge, registration to domain/host name 216.58.0.51
  122.  
  123. pbx*CLI> clear now
  124. REGISTER 11 headers, 0 lines
  125.  
  126. pbx*CLI> clear now
  127. Reliably Transmitting (NAT) to 216.58.0.51:5060:
  128. REGISTER sip:216.58.0.51 SIP/2.0
  129.  
  130. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK53949924;rport
  131.  
  132. Max-Forwards: 70
  133.  
  134. From: <sip:6477232824@216.58.0.51>;tag=as435931ef
  135.  
  136. To: <sip:6477232824@216.58.0.51>
  137.  
  138. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  139.  
  140. CSeq: 117 REGISTER
  141.  
  142. User-Agent: FPBX-2.9.0(1.6.2.15)
  143.  
  144. Authorization: Digest username="6477232824", realm="cia.com", algorithm=MD5, uri="sip:216.58.0.51", nonce="27ba261c", response="da80b0712107829f65f5704d745a5d1a"
  145.  
  146. Expires: 120
  147.  
  148. Contact: <sip:6477232824@99.227.42.4>
  149.  
  150. Content-Length: 0
  151.  
  152.  
  153.  
  154.  
  155. ---
  156.  
  157. pbx*CLI> clear now
  158. 
  159. <--- SIP read from UDP:216.58.0.51:5060 --->
  160. SIP/2.0 100 Trying
  161.  
  162. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK53949924;received=99.227.42.4;rport=5060
  163.  
  164. From: <sip:6477232824@216.58.0.51>;tag=as435931ef
  165.  
  166. To: <sip:6477232824@216.58.0.51>
  167.  
  168. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  169.  
  170. CSeq: 117 REGISTER
  171.  
  172. User-Agent: CIA.com PBX
  173.  
  174. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  175.  
  176. Supported: replaces
  177.  
  178. Contact: <sip:6477232824@216.58.0.51>
  179.  
  180. Content-Length: 0
  181.  
  182.  
  183.  
  184.  
  185. <------------->
  186.  
  187. pbx*CLI> clear now
  188. --- (11 headers 0 lines) ---
  189.  
  190. pbx*CLI> clear now
  191. 
  192. <--- SIP read from UDP:216.58.0.51:5060 --->
  193. SIP/2.0 200 OK
  194.  
  195. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK53949924;received=99.227.42.4;rport=5060
  196.  
  197. From: <sip:6477232824@216.58.0.51>;tag=as435931ef
  198.  
  199. To: <sip:6477232824@216.58.0.51>;tag=as76e4c03c
  200.  
  201. Call-ID: 20299b416b34c75714a0647e2982f8a9@127.0.0.1
  202.  
  203. CSeq: 117 REGISTER
  204.  
  205. User-Agent: CIA.com PBX
  206.  
  207. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  208.  
  209. Supported: replaces
  210.  
  211. Expires: 120
  212.  
  213. Contact: <sip:6477232824@99.227.42.4>;expires=120
  214.  
  215. Date: Sat, 27 Aug 2011 02:45:11 GMT
  216.  
  217. Content-Length: 0
  218.  
  219.  
  220.  
  221.  
  222. <------------->
  223. --- (13 headers 0 lines) ---
  224. Scheduling destruction of SIP dialog '20299b416b34c75714a0647e2982f8a9@127.0.0.1' in 32000 ms (Method: REGISTER)
  225.  
  226. pbx*CLI> clear now
  227.  
  228. pbx*CLI>
  229. No such command 'clear now' (type 'core show help clear now' for other possible commands)
  230.  
  231. pbx*CLI>
  232. 
  233. <--- SIP read from UDP:216.58.0.51:5060 --->
  234. INVITE sip:6477232824@99.227.42.4 SIP/2.0
  235.  
  236. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK5a9c80c1;rport
  237.  
  238. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  239.  
  240. To: <sip:6477232824@99.227.42.4>
  241.  
  242. Contact: <sip:4168853558@216.58.0.51>
  243.  
  244. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  245.  
  246. CSeq: 102 INVITE
  247.  
  248. User-Agent: CIA.com PBX
  249.  
  250. Max-Forwards: 70
  251.  
  252. Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
  253.  
  254. Date: Sat, 27 Aug 2011 02:45:20 GMT
  255.  
  256. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  257.  
  258. Supported: replaces
  259.  
  260. Content-Type: application/sdp
  261.  
  262. Content-Length: 285
  263.  
  264.  
  265.  
  266. v=0
  267.  
  268. o=root 22237 22237 IN IP4 216.58.0.51
  269.  
  270. s=session
  271.  
  272. c=IN IP4 216.58.0.51
  273.  
  274. t=0 0
  275.  
  276. m=audio 14192 RTP/AVP 0 8 3 101
  277.  
  278. a=rtpmap:0 PCMU/8000
  279.  
  280. a=rtpmap:8 PCMA/8000
  281.  
  282. a=rtpmap:3 GSM/8000
  283.  
  284. a=rtpmap:101 telephone-event/8000
  285.  
  286. a=fmtp:101 0-16
  287.  
  288. a=silenceSupp:off - - - -
  289.  
  290. a=ptime:20
  291.  
  292. a=sendrecv
  293.  
  294.  
  295. <------------->
  296. --- (15 headers 14 lines) ---
  297.  
  298. pbx*CLI>
  299.  == Using SIP RTP TOS bits 184
  300. == Using SIP RTP CoS mark 5
  301. Sending to 216.58.0.51 : 5060 (NAT)
  302. Using INVITE request as basis request - 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  303. Found peer '6477232824' for '4168853558' from 216.58.0.51:5060
  304.  
  305. pbx*CLI>
  306. Found RTP audio format 0
  307. Found RTP audio format 8
  308.  
  309. pbx*CLI>
  310. Found RTP audio format 3
  311.  
  312. pbx*CLI>
  313. Found RTP audio format 101
  314.  
  315. pbx*CLI>
  316. Found audio description format PCMU for ID 0
  317.  
  318. pbx*CLI>
  319. Found audio description format PCMA for ID 8
  320.  
  321. pbx*CLI>
  322. Found audio description format GSM for ID 3
  323.  
  324. pbx*CLI>
  325. Found audio description format telephone-event for ID 101
  326.  
  327. pbx*CLI>
  328. Capabilities: us - 0x4 (ulaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  329.  
  330. pbx*CLI>
  331. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  332.  
  333. pbx*CLI>
  334. Peer audio RTP is at port 216.58.0.51:14192
  335.  
  336. pbx*CLI>
  337. Looking for 6477232824 in from-trunk (domain 99.227.42.4)
  338.  
  339. pbx*CLI>
  340. list_route: hop: <sip:4168853558@216.58.0.51>
  341.  
  342. <--- Transmitting (NAT) to 216.58.0.51:5060 --->
  343. SIP/2.0 100 Trying
  344.  
  345. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK5a9c80c1;received=216.58.0.51;rport=5060
  346.  
  347. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  348.  
  349. To: <sip:6477232824@99.227.42.4>
  350.  
  351. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  352.  
  353. CSeq: 102 INVITE
  354.  
  355. Server: FPBX-2.9.0(1.6.2.15)
  356.  
  357. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  358.  
  359. Supported: replaces, timer
  360.  
  361. Contact: <sip:6477232824@99.227.42.4>
  362.  
  363. Content-Length: 0
  364.  
  365.  
  366.  
  367.  
  368. <------------>
  369. -- Executing [6477232824@from-trunk:1] Set("SIP/6477232824-00000022", "__FROM_DID=6477232824") in new stack
  370. -- Executing [6477232824@from-trunk:2] Gosub("SIP/6477232824-00000022", "app-blacklist-check,s,1") in new stack
  371. -- Executing [s@app-blacklist-check:1] GotoIf("SIP/6477232824-00000022", "0?blacklisted") in new stack
  372.  
  373. pbx*CLI>
  374.  -- Executing [s@app-blacklist-check:2] Set("SIP/6477232824-00000022", "CALLED_BLACKLIST=1") in new stack
  375. -- Executing [s@app-blacklist-check:3] Return("SIP/6477232824-00000022", "") in new stack
  376.  
  377. pbx*CLI>
  378.  -- Executing [6477232824@from-trunk:3] ExecIf("SIP/6477232824-00000022", "0 ?Set(CALLERID(name)=4168853558)") in new stack
  379.  
  380. pbx*CLI>
  381.  -- Executing [6477232824@from-trunk:4] Set("SIP/6477232824-00000022", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  382.  
  383. pbx*CLI>
  384.  -- Executing [6477232824@from-trunk:5] Set("SIP/6477232824-00000022", "CALLERPRES()=allowed_not_screened") in new stack
  385. -- Executing [6477232824@from-trunk:6] Goto("SIP/6477232824-00000022", "ext-group,601,1") in new stack
  386. -- Goto (ext-group,601,1)
  387.  
  388. pbx*CLI>
  389.  -- Executing [601@ext-group:1] Macro("SIP/6477232824-00000022", "user-callerid,") in new stack
  390.  
  391. pbx*CLI>
  392.  -- Executing [s@macro-user-callerid:1] Set("SIP/6477232824-00000022", "AMPUSER=4168853558") in new stack
  393.  
  394. pbx*CLI>
  395.  -- Executing [s@macro-user-callerid:2] GotoIf("SIP/6477232824-00000022", "0?report") in new stack
  396.  
  397. pbx*CLI>
  398.  -- Executing [s@macro-user-callerid:3] ExecIf("SIP/6477232824-00000022", "1?Set(REALCALLERIDNUM=4168853558)") in new stack
  399. -- Executing [s@macro-user-callerid:4] Set("SIP/6477232824-00000022", "AMPUSER=") in new stack
  400.  
  401. pbx*CLI>
  402.  -- Executing [s@macro-user-callerid:5] Set("SIP/6477232824-00000022", "AMPUSERCIDNAME=") in new stack
  403.  
  404. pbx*CLI>
  405.  -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6477232824-00000022", "1?report") in new stack
  406. -- Goto (macro-user-callerid,s,11)
  407.  
  408. pbx*CLI>
  409.  -- Executing [s@macro-user-callerid:11] GotoIf("SIP/6477232824-00000022", "0?continue") in new stack
  410.  
  411. pbx*CLI>
  412.  -- Executing [s@macro-user-callerid:12] Set("SIP/6477232824-00000022", "__TTL=64") in new stack
  413.  
  414. pbx*CLI>
  415.  -- Executing [s@macro-user-callerid:13] GotoIf("SIP/6477232824-00000022", "1?continue") in new stack
  416. -- Goto (macro-user-callerid,s,24)
  417.  
  418. pbx*CLI>
  419.  -- Executing [s@macro-user-callerid:24] Set("SIP/6477232824-00000022", "CALLERID(number)=4168853558") in new stack
  420.  
  421. pbx*CLI>
  422.  -- Executing [s@macro-user-callerid:25] Set("SIP/6477232824-00000022", "CALLERID(name)=Unknown") in new stack
  423.  
  424. pbx*CLI>
  425.  -- Executing [601@ext-group:2] Macro("SIP/6477232824-00000022", "blkvm-setifempty,") in new stack
  426.  
  427. pbx*CLI>
  428.  -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/6477232824-00000022", "1?init") in new stack
  429. -- Goto (macro-blkvm-setifempty,s,4)
  430.  
  431. pbx*CLI>
  432.  -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/6477232824-00000022", "__BLKVM_CHANNEL=SIP/6477232824-00000022") in new stack
  433.  
  434. pbx*CLI>
  435.  -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/6477232824-00000022", "SHARED(BLKVM,SIP/6477232824-00000022)=TRUE") in new stack
  436.  
  437. pbx*CLI>
  438.  -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/6477232824-00000022", "GOSUB_RETVAL=TRUE") in new stack
  439.  
  440. pbx*CLI>
  441.  -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/6477232824-00000022", "") in new stack
  442.  
  443. pbx*CLI>
  444.  -- Executing [601@ext-group:3] GotoIf("SIP/6477232824-00000022", "1?skipov") in new stack
  445.  
  446. pbx*CLI>
  447.  -- Goto (ext-group,601,6)
  448.  
  449. pbx*CLI>
  450.  -- Executing [601@ext-group:6] Set("SIP/6477232824-00000022", "RRNODEST=") in new stack
  451.  
  452. pbx*CLI>
  453.  -- Executing [601@ext-group:7] Set("SIP/6477232824-00000022", "__NODEST=601") in new stack
  454.  
  455. pbx*CLI>
  456.  -- Executing [601@ext-group:8] GosubIf("SIP/6477232824-00000022", "0?sub-rgsetcid,s,1") in new stack
  457.  
  458. pbx*CLI>
  459.  -- Executing [601@ext-group:9] Set("SIP/6477232824-00000022", "RecordMethod=Group") in new stack
  460.  
  461. pbx*CLI>
  462.  -- Executing [601@ext-group:10] Macro("SIP/6477232824-00000022", "record-enable,201-203,Group") in new stack
  463.  
  464. pbx*CLI>
  465.  -- Executing [s@macro-record-enable:1] GotoIf("SIP/6477232824-00000022", "1?check") in new stack
  466.  
  467. pbx*CLI>
  468.  -- Goto (macro-record-enable,s,4)
  469.  
  470. pbx*CLI>
  471.  -- Executing [s@macro-record-enable:4] ExecIf("SIP/6477232824-00000022", "0?MacroExit()") in new stack
  472.  
  473. pbx*CLI>
  474.  -- Executing [s@macro-record-enable:5] GotoIf("SIP/6477232824-00000022", "1?Group:OUT") in new stack
  475.  
  476. pbx*CLI>
  477.  -- Goto (macro-record-enable,s,6)
  478.  
  479. pbx*CLI>
  480.  -- Executing [s@macro-record-enable:6] Set("SIP/6477232824-00000022", "LOOPCNT=2") in new stack
  481.  
  482. pbx*CLI>
  483.  -- Executing [s@macro-record-enable:7] Set("SIP/6477232824-00000022", "ITER=1") in new stack
  484.  
  485. pbx*CLI>
  486.  -- Executing [s@macro-record-enable:8] GotoIf("SIP/6477232824-00000022", "1?continue") in new stack
  487.  
  488. pbx*CLI>
  489.  -- Goto (macro-record-enable,s,12)
  490.  
  491. pbx*CLI>
  492.  -- Executing [s@macro-record-enable:12] Set("SIP/6477232824-00000022", "ITER=2") in new stack
  493.  
  494. pbx*CLI>
  495.  -- Executing [s@macro-record-enable:13] GotoIf("SIP/6477232824-00000022", "1?begin") in new stack
  496.  
  497. pbx*CLI>
  498.  -- Goto (macro-record-enable,s,8)
  499.  
  500. pbx*CLI>
  501.  -- Executing [s@macro-record-enable:8] GotoIf("SIP/6477232824-00000022", "1?continue") in new stack
  502. -- Goto (macro-record-enable,s,12)
  503.  
  504. pbx*CLI>
  505.  -- Executing [s@macro-record-enable:12] Set("SIP/6477232824-00000022", "ITER=3") in new stack
  506.  
  507. pbx*CLI>
  508.  -- Executing [s@macro-record-enable:13] GotoIf("SIP/6477232824-00000022", "0?begin") in new stack
  509.  
  510. pbx*CLI>
  511.  -- Executing [s@macro-record-enable:14] GotoIf("SIP/6477232824-00000022", "0?IN") in new stack
  512.  
  513. pbx*CLI>
  514.  -- Executing [s@macro-record-enable:15] ExecIf("SIP/6477232824-00000022", "1?MacroExit()") in new stack
  515.  
  516. pbx*CLI>
  517.  -- Executing [601@ext-group:11] Set("SIP/6477232824-00000022", "RingGroupMethod=ringall") in new stack
  518.  
  519. pbx*CLI>
  520.  -- Executing [601@ext-group:12] Macro("SIP/6477232824-00000022", "dial,20,tr,201-203") in new stack
  521.  
  522. pbx*CLI>
  523.  -- Executing [s@macro-dial:1] GotoIf("SIP/6477232824-00000022", "1?dial") in new stack
  524.  
  525. pbx*CLI>
  526.  -- Goto (macro-dial,s,3)
  527. -- Executing [s@macro-dial:3] AGI("SIP/6477232824-00000022", "dialparties.agi") in new stack
  528.  
  529. pbx*CLI>
  530.  -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  531.  
  532. pbx*CLI>
  533.  dialparties.agi: Starting New Dialparties.agi
  534.  
  535. pbx*CLI>
  536.  dialparties.agi: Caller ID name is 'Unknown' number is '4168853558'
  537.  
  538. pbx*CLI>
  539.  > dialparties.agi: USE_CONFIRMATION: 'FALSE'
  540.  
  541. pbx*CLI>
  542.  > dialparties.agi: RINGGROUP_INDEX: ''
  543.  
  544. pbx*CLI>
  545.  dialparties.agi: Methodology of ring is 'ringall'
  546.  
  547. pbx*CLI>
  548.  -- dialparties.agi: Added extension 201 to extension map
  549.  
  550. pbx*CLI>
  551.  -- dialparties.agi: Added extension 203 to extension map
  552.  
  553. pbx*CLI>
  554.  -- dialparties.agi: Extension 201 cf is disabled
  555.  
  556. pbx*CLI>
  557.  -- dialparties.agi: Extension 203 cf is disabled
  558.  
  559. pbx*CLI>
  560.  -- dialparties.agi: Extension 201 do not disturb is disabled
  561.  
  562. pbx*CLI>
  563.  -- dialparties.agi: Extension 203 do not disturb is disabled
  564.  
  565. pbx*CLI>
  566.  > dialparties.agi: extnum 201 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  567.  
  568. pbx*CLI>
  569.  -- dialparties.agi: dbset CALLTRACE/201 to 4168853558
  570.  
  571. pbx*CLI>
  572.  > dialparties.agi: extnum 203 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  573.  
  574. pbx*CLI>
  575.  -- dialparties.agi: dbset CALLTRACE/203 to 4168853558
  576.  
  577. pbx*CLI>
  578.  -- dialparties.agi: Filtered ARG3: 201-203
  579.  
  580. pbx*CLI>
  581.  > dialparties.agi: NODEST: 601 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
  582.  
  583. pbx*CLI>
  584.  > dialparties.agi: NODEST: 601 blkvm enabled macro already in dialopts: trM(auto-blkvm)
  585.  
  586. pbx*CLI>
  587.  -- <SIP/6477232824-00000022>AGI Script dialparties.agi completed, returning 0
  588.  
  589. pbx*CLI>
  590.  -- Executing [s@macro-dial:7] Dial("SIP/6477232824-00000022", "SIP/201&SIP/203,20,trM(auto-blkvm)") in new stack
  591.  
  592. pbx*CLI>
  593.  == Using SIP RTP TOS bits 184
  594.  
  595. pbx*CLI>
  596.  == Using SIP RTP CoS mark 5
  597.  
  598. pbx*CLI>
  599.  == Using SIP RTP TOS bits 184
  600. == Using SIP RTP CoS mark 5
  601.  
  602. pbx*CLI>
  603.  -- Called 203
  604.  
  605. pbx*CLI>
  606. 
  607. <--- Transmitting (NAT) to 216.58.0.51:5060 --->
  608. SIP/2.0 180 Ringing
  609.  
  610. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK5a9c80c1;received=216.58.0.51;rport=5060
  611.  
  612. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  613.  
  614. To: <sip:6477232824@99.227.42.4>;tag=as227226aa
  615.  
  616. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  617.  
  618. CSeq: 102 INVITE
  619.  
  620. Server: FPBX-2.9.0(1.6.2.15)
  621.  
  622. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  623.  
  624. Supported: replaces, timer
  625.  
  626. Contact: <sip:6477232824@99.227.42.4>
  627.  
  628. Content-Length: 0
  629.  
  630.  
  631.  
  632.  
  633. <------------>
  634.  
  635. pbx*CLI>
  636.  -- SIP/203-00000023 is ringing
  637.  
  638. pbx*CLI>
  639. 
  640. <--- Transmitting (NAT) to 216.58.0.51:5060 --->
  641. SIP/2.0 180 Ringing
  642.  
  643. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK5a9c80c1;received=216.58.0.51;rport=5060
  644.  
  645. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  646.  
  647. To: <sip:6477232824@99.227.42.4>;tag=as227226aa
  648.  
  649. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  650.  
  651. CSeq: 102 INVITE
  652.  
  653. Server: FPBX-2.9.0(1.6.2.15)
  654.  
  655. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  656.  
  657. Supported: replaces, timer
  658.  
  659. Contact: <sip:6477232824@99.227.42.4>
  660.  
  661. Content-Length: 0
  662.  
  663.  
  664.  
  665.  
  666. <------------>
  667.  
  668. pbx*CLI>
  669.  -- SIP/203-00000023 answered SIP/6477232824-00000022
  670.  
  671. pbx*CLI>
  672.  -- Executing [s@macro-auto-blkvm:1] Set("SIP/203-00000023", "__MACRO_RESULT=") in new stack
  673.  
  674. pbx*CLI>
  675.  -- Executing [s@macro-auto-blkvm:2] Macro("SIP/203-00000023", "blkvm-clr,") in new stack
  676.  
  677. pbx*CLI>
  678.  -- Executing [s@macro-blkvm-clr:1] Set("SIP/203-00000023", "SHARED(BLKVM,SIP/6477232824-00000022)=") in new stack
  679.  
  680. pbx*CLI>
  681.  -- Executing [s@macro-blkvm-clr:2] Set("SIP/203-00000023", "GOSUB_RETVAL=") in new stack
  682.  
  683. pbx*CLI>
  684.  -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/203-00000023", "") in new stack
  685.  
  686. pbx*CLI>
  687. Audio is at 99.227.42.4 port 20006
  688. Adding codec 0x4 (ulaw) to SDP
  689. Adding non-codec 0x1 (telephone-event) to SDP
  690.  
  691. pbx*CLI>
  692. 
  693. <--- Reliably Transmitting (NAT) to 216.58.0.51:5060 --->
  694. SIP/2.0 200 OK
  695.  
  696. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK5a9c80c1;received=216.58.0.51;rport=5060
  697.  
  698. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  699.  
  700. To: <sip:6477232824@99.227.42.4>;tag=as227226aa
  701.  
  702. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  703.  
  704. CSeq: 102 INVITE
  705.  
  706. Server: FPBX-2.9.0(1.6.2.15)
  707.  
  708. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  709.  
  710. Supported: replaces, timer
  711.  
  712. Contact: <sip:6477232824@99.227.42.4>
  713.  
  714. Content-Type: application/sdp
  715.  
  716. Content-Length: 235
  717.  
  718.  
  719.  
  720. v=0
  721.  
  722. o=root 2146043800 2146043800 IN IP4 99.227.42.4
  723.  
  724. s=Asterisk PBX 1.6.2.15
  725.  
  726. c=IN IP4 99.227.42.4
  727.  
  728. t=0 0
  729.  
  730. m=audio 20006 RTP/AVP 0 101
  731.  
  732. a=rtpmap:0 PCMU/8000
  733.  
  734. a=rtpmap:101 telephone-event/8000
  735.  
  736. a=fmtp:101 0-16
  737.  
  738. a=ptime:20
  739.  
  740. a=sendrecv
  741.  
  742.  
  743. <------------>
  744.  
  745. pbx*CLI>
  746. 
  747. <--- SIP read from UDP:216.58.0.51:5060 --->
  748. ACK sip:6477232824@99.227.42.4 SIP/2.0
  749.  
  750. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK661fd855;rport
  751.  
  752. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  753.  
  754. To: <sip:6477232824@99.227.42.4>;tag=as227226aa
  755.  
  756. Contact: <sip:4168853558@216.58.0.51>
  757.  
  758. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  759.  
  760. CSeq: 102 ACK
  761.  
  762. User-Agent: CIA.com PBX
  763.  
  764. Max-Forwards: 70
  765.  
  766. Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
  767.  
  768. Content-Length: 0
  769.  
  770.  
  771. pbx*CLI>
  772. 
  773.  
  774.  
  775. <------------->
  776. --- (11 headers 0 lines) ---
  777.  
  778. pbx*CLI>
  779. Really destroying SIP dialog '20299b416b34c75714a0647e2982f8a9@127.0.0.1' Method: REGISTER
  780.  
  781. pbx*CLI>
  782.  -- Executing [h@macro-dial:1] Macro("SIP/6477232824-00000022", "hangupcall") in new stack
  783. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/6477232824-00000022", "1?theend") in new stack
  784. -- Goto (macro-hangupcall,s,3)
  785. -- Executing [s@macro-hangupcall:3] Hangup("SIP/6477232824-00000022", "") in new stack
  786. == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/6477232824-00000022' in macro 'hangupcall'
  787. == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/6477232824-00000022' in macro 'dial'
  788. == Spawn extension (ext-group, 601, 12) exited non-zero on 'SIP/6477232824-00000022'
  789. Scheduling destruction of SIP dialog '5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51' in 6400 ms (Method: ACK)
  790. set_destination: Parsing <sip:4168853558@216.58.0.51> for address/port to send to
  791. set_destination: set destination to 216.58.0.51, port 5060
  792. Reliably Transmitting (NAT) to 216.58.0.51:5060:
  793. BYE sip:4168853558@216.58.0.51 SIP/2.0
  794.  
  795. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK6808d864;rport
  796.  
  797. Max-Forwards: 70
  798.  
  799. From: <sip:6477232824@99.227.42.4>;tag=as227226aa
  800.  
  801. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  802.  
  803. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  804.  
  805. CSeq: 102 BYE
  806.  
  807. User-Agent: FPBX-2.9.0(1.6.2.15)
  808.  
  809. X-Asterisk-HangupCause: Normal Clearing
  810.  
  811. X-Asterisk-HangupCauseCode: 16
  812.  
  813. Content-Length: 0
  814.  
  815.  
  816.  
  817.  
  818. ---
  819.  
  820. pbx*CLI>
  821. Retransmitting #1 (NAT) to 216.58.0.51:5060:
  822. BYE sip:4168853558@216.58.0.51 SIP/2.0
  823.  
  824. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK6808d864;rport
  825.  
  826. Max-Forwards: 70
  827.  
  828. From: <sip:6477232824@99.227.42.4>;tag=as227226aa
  829.  
  830. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  831.  
  832. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  833.  
  834. CSeq: 102 BYE
  835.  
  836. User-Agent: FPBX-2.9.0(1.6.2.15)
  837.  
  838. X-Asterisk-HangupCause: Normal Clearing
  839.  
  840. X-Asterisk-HangupCauseCode: 16
  841.  
  842. Content-Length: 0
  843.  
  844.  
  845.  
  846.  
  847. ---
  848.  
  849. pbx*CLI>
  850. 
  851. <--- SIP read from UDP:216.58.0.51:5060 --->
  852. SIP/2.0 200 OK
  853.  
  854. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK6808d864;received=99.227.42.4;rport=5060
  855.  
  856. From: <sip:6477232824@99.227.42.4>;tag=as227226aa
  857.  
  858. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  859.  
  860. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  861.  
  862. CSeq: 102 BYE
  863.  
  864. User-Agent: CIA.com PBX
  865.  
  866. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  867.  
  868. Supported: replaces
  869.  
  870. Contact: <sip:4168853558@216.58.0.51>
  871.  
  872. Content-Length: 0
  873.  
  874.  
  875.  
  876.  
  877. <------------->
  878. --- (11 headers 0 lines) ---
  879. SIP Response message for INCOMING dialog BYE arrived
  880. Really destroying SIP dialog '5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51' Method: ACK
  881.  
  882. pbx*CLI>
  883. 
  884. <--- SIP read from UDP:216.58.0.51:5060 --->
  885. SIP/2.0 200 OK
  886.  
  887. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK6808d864;received=99.227.42.4;rport=5060
  888.  
  889. From: <sip:6477232824@99.227.42.4>;tag=as227226aa
  890.  
  891. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as23a7ff6b
  892.  
  893. Call-ID: 5ef2083c2b7e3f841383b49a4c4d4ce1@216.58.0.51
  894.  
  895. CSeq: 102 BYE
  896.  
  897. User-Agent: CIA.com PBX
  898.  
  899. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  900.  
  901. Supported: replaces
  902.  
  903. Contact: <sip:4168853558@216.58.0.51>
  904.  
  905. Content-Length: 0
  906.  
  907.  
  908.  
  909.  
  910. <------------->
  911.  
  912. pbx*CLI>
  913. --- (11 headers 0 lines) ---
  914.  
  915. pbx*CLI>
  916. Reliably Transmitting (NAT) to 216.58.0.51:5060:
  917. OPTIONS sip:216.58.0.51 SIP/2.0
  918.  
  919. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK1ae317a3;rport
  920.  
  921. Max-Forwards: 70
  922.  
  923. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as647be0e6
  924.  
  925. To: <sip:216.58.0.51>
  926.  
  927. Contact: <sip:Unknown@99.227.42.4>
  928.  
  929. Call-ID: 3f9b78f5179221d765f026352bf38943@99.227.42.4
  930.  
  931. CSeq: 102 OPTIONS
  932.  
  933. User-Agent: FPBX-2.9.0(1.6.2.15)
  934.  
  935. Date: Sat, 27 Aug 2011 02:27:27 GMT
  936.  
  937. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  938.  
  939. Supported: replaces, timer
  940.  
  941. Content-Length: 0
  942.  
  943.  
  944.  
  945.  
  946. ---
  947.  
  948. pbx*CLI>
  949. 
  950. <--- SIP read from UDP:216.58.0.51:5060 --->
  951. SIP/2.0 404 Not Found
  952.  
  953. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK1ae317a3;received=99.227.42.4;rport=5060
  954.  
  955. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as647be0e6
  956.  
  957. To: <sip:216.58.0.51>;tag=as70df2000
  958.  
  959. Call-ID: 3f9b78f5179221d765f026352bf38943@99.227.42.4
  960.  
  961. CSeq: 102 OPTIONS
  962.  
  963. User-Agent: CIA.com PBX
  964.  
  965. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  966.  
  967. Supported: replaces
  968.  
  969. Accept: application/sdp
  970.  
  971. Content-Length: 0
  972.  
  973.  
  974.  
  975.  
  976. <------------->
  977. --- (11 headers 0 lines) ---
  978. Really destroying SIP dialog '3f9b78f5179221d765f026352bf38943@99.227.42.4' Method: OPTIONS
  979.  
  980. pbx*CLI> exit
  981.  
  982. Executing last minute cleanups
  983. ]0;root@pbx:~[root@pbx ~]# exit
  984. logout
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