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  1. =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.08.29 13:27:14 =~=~=~=~=~=~=~=~=~=~=~=
  2. login as: root
  3. root@pbx.hecint.com's password:
  4. Last login: Sun Aug 28 20:53:21 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 = 13167)
  16. pbx*CLI>
  17. Verbosity was 3 and is now 7
  18.  
  19. pbx*CLI>
  20.  == Manager 'admin' logged on from 127.0.0.1
  21.  
  22. pbx*CLI> sip set
  23.  == Manager 'admin' logged off from 127.0.0.1
  24.  
  25. pbx*CLI> sip set debug on
  26.  
  27. pbx*CLI>
  28. SIP Debugging enabled
  29.  
  30. pbx*CLI>
  31. 
  32. <--- SIP read from UDP:10.10.10.12:5060 --->
  33. INVITE sip:4162352999@10.10.10.7 SIP/2.0
  34.  
  35. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-2861715a
  36.  
  37. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  38.  
  39. To: <sip:4162352999@10.10.10.7>
  40.  
  41. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  42.  
  43. CSeq: 101 INVITE
  44.  
  45. Max-Forwards: 70
  46.  
  47. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  48.  
  49. Expires: 240
  50.  
  51. User-Agent: Linksys/SPA942-6.1.5(a)
  52.  
  53. Content-Length: 208
  54.  
  55. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  56.  
  57. Supported: replaces
  58.  
  59. Content-Type: application/sdp
  60.  
  61.  
  62.  
  63. v=0
  64.  
  65. o=- 21610799 21610799 IN IP4 10.10.10.12
  66.  
  67. s=-
  68.  
  69. c=IN IP4 10.10.10.12
  70.  
  71. t=0 0
  72.  
  73. m=audio 13531 RTP/AVP 0 101
  74.  
  75. a=rtpmap:0 PCMU/8000
  76.  
  77. a=rtpmap:101 telephone-event/8000
  78.  
  79. a=fmtp:101 0-15
  80.  
  81. a=ptime:30
  82.  
  83. a=sendrecv
  84.  
  85.  
  86. <------------->
  87. --- (14 headers 11 lines) ---
  88.  
  89. pbx*CLI>
  90.  == Using SIP RTP TOS bits 184
  91. == Using SIP RTP CoS mark 5
  92.  
  93. pbx*CLI>
  94. Sending to 10.10.10.12 : 5060 (NAT)
  95.  
  96. pbx*CLI>
  97. Using INVITE request as basis request - 85ec10bc-d355ba22@10.10.10.12
  98.  
  99. pbx*CLI>
  100. Found peer '203' for '203' from 10.10.10.12:5060
  101.  
  102. pbx*CLI>
  103. 
  104. <--- Reliably Transmitting (NAT) to 10.10.10.12:5060 --->
  105. SIP/2.0 401 Unauthorized
  106.  
  107. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-2861715a;received=10.10.10.12
  108.  
  109. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  110.  
  111. To: <sip:4162352999@10.10.10.7>;tag=as0c9c29b9
  112.  
  113. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  114.  
  115. CSeq: 101 INVITE
  116.  
  117. Server: FPBX-2.9.0(1.6.2.15)
  118.  
  119. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  120.  
  121. Supported: replaces, timer
  122.  
  123. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f6212ab"
  124.  
  125. Content-Length: 0
  126.  
  127.  
  128.  
  129.  
  130. <------------>
  131.  
  132. pbx*CLI>
  133. Scheduling destruction of SIP dialog '85ec10bc-d355ba22@10.10.10.12' in 6400 ms (Method: INVITE)
  134.  
  135. pbx*CLI>
  136. 
  137. <--- SIP read from UDP:10.10.10.12:5060 --->
  138. ACK sip:4162352999@10.10.10.7 SIP/2.0
  139.  
  140. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-2861715a
  141.  
  142. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  143.  
  144. To: <sip:4162352999@10.10.10.7>;tag=as0c9c29b9
  145.  
  146. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  147.  
  148. CSeq: 101 ACK
  149.  
  150. Max-Forwards: 70
  151.  
  152. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  153.  
  154. User-Agent: Linksys/SPA942-6.1.5(a)
  155.  
  156. Content-Length: 0
  157.  
  158.  
  159.  
  160.  
  161. <------------->
  162. --- (10 headers 0 lines) ---
  163.  
  164. pbx*CLI>
  165. 
  166. <--- SIP read from UDP:10.10.10.12:5060 --->
  167. INVITE sip:4162352999@10.10.10.7 SIP/2.0
  168.  
  169. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a
  170.  
  171. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  172.  
  173. To: <sip:4162352999@10.10.10.7>
  174.  
  175. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  176.  
  177. CSeq: 102 INVITE
  178.  
  179. Max-Forwards: 70
  180.  
  181. Authorization: Digest username="203",realm="asterisk",nonce="7f6212ab",uri="sip:4162352999@10.10.10.7",algorithm=MD5,response="20ca43714dd1f9007c4b7f6077299b2d"
  182.  
  183. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  184.  
  185. Expires: 240
  186.  
  187. User-Agent: Linksys/SPA942-6.1.5(a)
  188.  
  189. Content-Length: 208
  190.  
  191. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  192.  
  193. Supported: replaces
  194.  
  195. Content-Type: application/sdp
  196.  
  197.  
  198.  
  199. v=0
  200.  
  201. o=- 21610799 21610799 IN IP4 10.10.10.12
  202.  
  203. s=-
  204.  
  205. c=IN IP4 10.10.10.12
  206.  
  207. t=0 0
  208.  
  209. m=audio 13531 RTP/AVP 0 101
  210.  
  211. a=rtpmap:0 PCMU/8000
  212.  
  213. a=rtpmap:101 telephone-event/8000
  214.  
  215. a=fmtp:101 0-15
  216.  
  217. a=ptime:30
  218.  
  219. a=sendrecv
  220.  
  221.  
  222. <------------->
  223. --- (15 headers 11 lines) ---
  224. Sending to 10.10.10.12 : 5060 (NAT)
  225. Using INVITE request as basis request - 85ec10bc-d355ba22@10.10.10.12
  226. Found peer '203' for '203' from 10.10.10.12:5060
  227.  
  228. pbx*CLI>
  229. Found RTP audio format 0
  230. Found RTP audio format 101
  231. Found audio description format PCMU for ID 0
  232.  
  233. pbx*CLI>
  234. Found audio description format telephone-event for ID 101
  235.  
  236. pbx*CLI>
  237. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  238. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  239. Peer audio RTP is at port 10.10.10.12:13531
  240. Looking for 4162352999 in from-internal (domain 10.10.10.7)
  241.  
  242. pbx*CLI>
  243. list_route: hop: <sip:203@10.10.10.12:5060>
  244.  
  245. <--- Transmitting (NAT) to 10.10.10.12:5060 --->
  246. SIP/2.0 100 Trying
  247.  
  248. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a;received=10.10.10.12
  249.  
  250. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  251.  
  252. To: <sip:4162352999@10.10.10.7>
  253.  
  254. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  255.  
  256. CSeq: 102 INVITE
  257.  
  258. Server: FPBX-2.9.0(1.6.2.15)
  259.  
  260. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  261.  
  262. Supported: replaces, timer
  263.  
  264. Contact: <sip:4162352999@10.10.10.7>
  265.  
  266. Content-Length: 0
  267.  
  268.  
  269.  
  270.  
  271. <------------>
  272.  
  273. pbx*CLI>
  274.  -- Executing [4162352999@from-internal:1] Macro("SIP/203-00000014", "user-callerid,LIMIT,") in new stack
  275.  
  276. pbx*CLI>
  277.  -- Executing [s@macro-user-callerid:1] Set("SIP/203-00000014", "AMPUSER=203") in new stack
  278.  
  279. pbx*CLI>
  280.  -- Executing [s@macro-user-callerid:2] GotoIf("SIP/203-00000014", "0?report") in new stack
  281.  
  282. pbx*CLI>
  283.  -- Executing [s@macro-user-callerid:3] ExecIf("SIP/203-00000014", "1?Set(REALCALLERIDNUM=203)") in new stack
  284.  
  285. pbx*CLI>
  286.  -- Executing [s@macro-user-callerid:4] Set("SIP/203-00000014", "AMPUSER=203") in new stack
  287.  
  288. pbx*CLI>
  289.  -- Executing [s@macro-user-callerid:5] Set("SIP/203-00000014", "AMPUSERCIDNAME=Itamar Desk") in new stack
  290.  
  291. pbx*CLI>
  292.  -- Executing [s@macro-user-callerid:6] GotoIf("SIP/203-00000014", "0?report") in new stack
  293.  
  294. pbx*CLI>
  295.  -- Executing [s@macro-user-callerid:7] Set("SIP/203-00000014", "AMPUSERCID=203") in new stack
  296.  
  297. pbx*CLI>
  298.  -- Executing [s@macro-user-callerid:8] Set("SIP/203-00000014", "CALLERID(all)="Itamar Desk" <203>") in new stack
  299.  
  300. pbx*CLI>
  301.  -- Executing [s@macro-user-callerid:9] GotoIf("SIP/203-00000014", "0?limit") in new stack
  302.  
  303. pbx*CLI>
  304.  -- Executing [s@macro-user-callerid:10] ExecIf("SIP/203-00000014", "1?Set(GROUP(concurrency_limit)=203)") in new stack
  305.  
  306. pbx*CLI>
  307.  -- Executing [s@macro-user-callerid:11] GotoIf("SIP/203-00000014", "1?continue") in new stack
  308.  
  309. pbx*CLI>
  310.  -- Goto (macro-user-callerid,s,24)
  311.  
  312. pbx*CLI>
  313.  -- Executing [s@macro-user-callerid:24] Set("SIP/203-00000014", "CALLERID(number)=203") in new stack
  314.  
  315. pbx*CLI>
  316.  -- Executing [s@macro-user-callerid:25] Set("SIP/203-00000014", "CALLERID(name)=Itamar Desk") in new stack
  317.  
  318. pbx*CLI>
  319.  -- Executing [4162352999@from-internal:2] Set("SIP/203-00000014", "MOHCLASS=default") in new stack
  320.  
  321. pbx*CLI>
  322.  -- Executing [4162352999@from-internal:3] Set("SIP/203-00000014", "_NODEST=") in new stack
  323.  
  324. pbx*CLI>
  325.  -- Executing [4162352999@from-internal:4] Macro("SIP/203-00000014", "record-enable,203,OUT,") in new stack
  326.  
  327. pbx*CLI>
  328.  -- Executing [s@macro-record-enable:1] GotoIf("SIP/203-00000014", "1?check") in new stack
  329.  
  330. pbx*CLI>
  331.  -- Goto (macro-record-enable,s,4)
  332.  
  333. pbx*CLI>
  334.  -- Executing [s@macro-record-enable:4] ExecIf("SIP/203-00000014", "0?MacroExit()") in new stack
  335.  
  336. pbx*CLI>
  337.  -- Executing [s@macro-record-enable:5] GotoIf("SIP/203-00000014", "0?Group:OUT") in new stack
  338.  
  339. pbx*CLI>
  340.  -- Goto (macro-record-enable,s,14)
  341.  
  342. pbx*CLI>
  343.  == Manager 'admin' logged on from 127.0.0.1
  344.  
  345. pbx*CLI>
  346.  -- Executing [s@macro-record-enable:14] GotoIf("SIP/203-00000014", "0?IN") in new stack
  347.  
  348. pbx*CLI>
  349.  -- Executing [s@macro-record-enable:15] ExecIf("SIP/203-00000014", "1?MacroExit()") in new stack
  350.  
  351. pbx*CLI>
  352.  -- Executing [4162352999@from-internal:5] Macro("SIP/203-00000014", "dialout-trunk,1,4162352999,") in new stack
  353.  
  354. pbx*CLI>
  355.  -- Executing [s@macro-dialout-trunk:1] Set("SIP/203-00000014", "DIAL_TRUNK=1") in new stack
  356.  
  357. pbx*CLI>
  358.  -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/203-00000014", "0?sub-pincheck,s,1") in new stack
  359.  
  360. pbx*CLI>
  361.  -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/203-00000014", "0?disabletrunk,1") in new stack
  362.  
  363. pbx*CLI>
  364.  -- Executing [s@macro-dialout-trunk:4] Set("SIP/203-00000014", "DIAL_NUMBER=4162352999") in new stack
  365.  
  366. pbx*CLI>
  367.  -- Executing [s@macro-dialout-trunk:5] Set("SIP/203-00000014", "DIAL_TRUNK_OPTIONS=tr") in new stack
  368.  
  369. pbx*CLI>
  370.  -- Executing [s@macro-dialout-trunk:6] Set("SIP/203-00000014", "OUTBOUND_GROUP=OUT_1") in new stack
  371.  
  372. pbx*CLI>
  373.  -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/203-00000014", "1?nomax") in new stack
  374.  
  375. pbx*CLI>
  376.  -- Goto (macro-dialout-trunk,s,9)
  377.  
  378. pbx*CLI>
  379.  -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/203-00000014", "0?skipoutcid") in new stack
  380.  
  381. pbx*CLI>
  382.  -- Executing [s@macro-dialout-trunk:10] Set("SIP/203-00000014", "DIAL_TRUNK_OPTIONS=") in new stack
  383.  
  384. pbx*CLI>
  385.  -- Executing [s@macro-dialout-trunk:11] Macro("SIP/203-00000014", "outbound-callerid,1") in new stack
  386.  
  387. pbx*CLI>
  388.  -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/203-00000014", "0?Set(CALLERPRES()=)") in new stack
  389.  
  390. pbx*CLI>
  391.  -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/203-00000014", "0?Set(REALCALLERIDNUM=203)") in new stack
  392.  
  393. pbx*CLI>
  394.  -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/203-00000014", "1?normcid") in new stack
  395.  
  396. pbx*CLI>
  397.  -- Goto (macro-outbound-callerid,s,6)
  398.  
  399. pbx*CLI>
  400.  -- Executing [s@macro-outbound-callerid:6] Set("SIP/203-00000014", "USEROUTCID=") in new stack
  401.  
  402. pbx*CLI>
  403.  -- Executing [s@macro-outbound-callerid:7] Set("SIP/203-00000014", "EMERGENCYCID=") in new stack
  404.  
  405. pbx*CLI>
  406.  -- Executing [s@macro-outbound-callerid:8] Set("SIP/203-00000014", "TRUNKOUTCID=<6477232824>") in new stack
  407.  
  408. pbx*CLI>
  409.  -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/203-00000014", "1?trunkcid") in new stack
  410.  
  411. pbx*CLI>
  412.  -- Goto (macro-outbound-callerid,s,12)
  413.  
  414. pbx*CLI>
  415.  -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/203-00000014", "1?Set(CALLERID(all)=<6477232824>)") in new stack
  416.  
  417. pbx*CLI>
  418.  -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/203-00000014", "0?Set(CALLERID(all)=)") in new stack
  419.  
  420. pbx*CLI>
  421.  -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/203-00000014", "1?Set(CALLERID(all)=<6477232824>)") in new stack
  422.  
  423. pbx*CLI>
  424.  -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/203-00000014", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  425.  
  426. pbx*CLI>
  427.  -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/203-00000014", "1?sub-flp-1,s,1") in new stack
  428.  
  429. pbx*CLI>
  430.  -- Executing [s@sub-flp-1:1] ExecIf("SIP/203-00000014", "0?Return()") in new stack
  431.  
  432. pbx*CLI>
  433.  -- Executing [s@sub-flp-1:2] ExecIf("SIP/203-00000014", "1?Set(TARGET_FLP_1=14162352999)") in new stack
  434.  
  435. pbx*CLI>
  436.  -- Executing [s@sub-flp-1:3] GotoIf("SIP/203-00000014", "1?match") in new stack
  437.  
  438. pbx*CLI>
  439.  -- Goto (sub-flp-1,s,8)
  440.  
  441. pbx*CLI>
  442.  -- Executing [s@sub-flp-1:8] Set("SIP/203-00000014", "DIAL_NUMBER=14162352999") in new stack
  443.  
  444. pbx*CLI>
  445.  -- Executing [s@sub-flp-1:9] Return("SIP/203-00000014", "") in new stack
  446.  
  447. pbx*CLI>
  448.  -- Executing [s@macro-dialout-trunk:13] Set("SIP/203-00000014", "OUTNUM=14162352999") in new stack
  449.  
  450. pbx*CLI>
  451.  -- Executing [s@macro-dialout-trunk:14] Set("SIP/203-00000014", "custom=SIP/6477232824") in new stack
  452.  
  453. pbx*CLI>
  454.  -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/203-00000014", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
  455.  
  456. pbx*CLI>
  457.  -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/203-00000014", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
  458.  
  459. pbx*CLI>
  460.  -- Executing [s@macro-dialout-trunk:17] Macro("SIP/203-00000014", "dialout-trunk-predial-hook,") in new stack
  461.  
  462. pbx*CLI>
  463.  -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/203-00000014", "") in new stack
  464.  
  465. pbx*CLI>
  466.  -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/203-00000014", "0?bypass,1") in new stack
  467.  
  468. pbx*CLI>
  469.  -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/203-00000014", "0?customtrunk") in new stack
  470.  
  471. pbx*CLI>
  472.  -- Executing [s@macro-dialout-trunk:20] Dial("SIP/203-00000014", "SIP/6477232824/14162352999,300,") in new stack
  473.  
  474. pbx*CLI>
  475.  == Using SIP RTP TOS bits 184
  476.  
  477. pbx*CLI>
  478.  == Using SIP RTP CoS mark 5
  479.  
  480. pbx*CLI>
  481. Audio is at 99.227.42.4 port 20004
  482.  
  483. pbx*CLI>
  484. Adding codec 0x4 (ulaw) to SDP
  485.  
  486. pbx*CLI>
  487. Adding non-codec 0x1 (telephone-event) to SDP
  488.  
  489. pbx*CLI>
  490. Reliably Transmitting (no NAT) to 10.10.10.2:5060:
  491. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  492.  
  493. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  494.  
  495. Max-Forwards: 70
  496.  
  497. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  498.  
  499. To: <sip:14162352999@216.58.0.51>
  500.  
  501. Contact: <sip:6477232824@99.227.42.4>
  502.  
  503. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  504.  
  505. CSeq: 102 INVITE
  506.  
  507. User-Agent: FPBX-2.9.0(1.6.2.15)
  508.  
  509. Date: Mon, 29 Aug 2011 17:09:06 GMT
  510.  
  511. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  512.  
  513. Supported: replaces, timer
  514.  
  515. Content-Type: application/sdp
  516.  
  517. Content-Length: 233
  518.  
  519.  
  520.  
  521. v=0
  522.  
  523. o=root 660440603 660440603 IN IP4 99.227.42.4
  524.  
  525. s=Asterisk PBX 1.6.2.15
  526.  
  527. c=IN IP4 99.227.42.4
  528.  
  529. t=0 0
  530.  
  531. m=audio 20004 RTP/AVP 0 101
  532.  
  533. a=rtpmap:0 PCMU/8000
  534.  
  535. a=rtpmap:101 telephone-event/8000
  536.  
  537. a=fmtp:101 0-16
  538.  
  539. a=ptime:20
  540.  
  541. a=sendrecv
  542.  
  543.  
  544. ---
  545.  
  546. pbx*CLI>
  547.  -- Called 6477232824/14162352999
  548.  
  549. pbx*CLI>
  550. Retransmitting #1 (no NAT) to 10.10.10.2:5060:
  551. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  552.  
  553. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  554.  
  555. Max-Forwards: 70
  556.  
  557. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  558.  
  559. To: <sip:14162352999@216.58.0.51>
  560.  
  561. Contact: <sip:6477232824@99.227.42.4>
  562.  
  563. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  564.  
  565. CSeq: 102 INVITE
  566.  
  567. User-Agent: FPBX-2.9.0(1.6.2.15)
  568.  
  569. Date: Mon, 29 Aug 2011 17:09:06 GMT
  570.  
  571. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  572.  
  573. Supported: replaces, timer
  574.  
  575. Content-Type: application/sdp
  576.  
  577. Content-Length: 233
  578.  
  579.  
  580.  
  581. v=0
  582.  
  583. o=root 660440603 660440603 IN IP4 99.227.42.4
  584.  
  585. s=Asterisk PBX 1.6.2.15
  586.  
  587. c=IN IP4 99.227.42.4
  588.  
  589. t=0 0
  590.  
  591. m=audio 20004 RTP/AVP 0 101
  592.  
  593. a=rtpmap:0 PCMU/8000
  594.  
  595. a=rtpmap:101 telephone-event/8000
  596.  
  597. a=fmtp:101 0-16
  598.  
  599. a=ptime:20
  600.  
  601. a=sendrecv
  602.  
  603.  
  604. ---
  605.  
  606. pbx*CLI>
  607. Retransmitting #2 (no NAT) to 10.10.10.2:5060:
  608. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  609.  
  610. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  611.  
  612. Max-Forwards: 70
  613.  
  614. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  615.  
  616. To: <sip:14162352999@216.58.0.51>
  617.  
  618. Contact: <sip:6477232824@99.227.42.4>
  619.  
  620. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  621.  
  622. CSeq: 102 INVITE
  623.  
  624. User-Agent: FPBX-2.9.0(1.6.2.15)
  625.  
  626. Date: Mon, 29 Aug 2011 17:09:06 GMT
  627.  
  628. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  629.  
  630. Supported: replaces, timer
  631.  
  632. Content-Type: application/sdp
  633.  
  634. Content-Length: 233
  635.  
  636.  
  637.  
  638. v=0
  639.  
  640. o=root 660440603 660440603 IN IP4 99.227.42.4
  641.  
  642. s=Asterisk PBX 1.6.2.15
  643.  
  644. c=IN IP4 99.227.42.4
  645.  
  646. t=0 0
  647.  
  648. m=audio 20004 RTP/AVP 0 101
  649.  
  650. a=rtpmap:0 PCMU/8000
  651.  
  652. a=rtpmap:101 telephone-event/8000
  653.  
  654. a=fmtp:101 0-16
  655.  
  656. a=ptime:20
  657.  
  658. a=sendrecv
  659.  
  660.  
  661. ---
  662.  
  663. pbx*CLI>
  664.  == Manager 'admin' logged off from 127.0.0.1
  665.  
  666. pbx*CLI>
  667. Retransmitting #3 (no NAT) to 10.10.10.2:5060:
  668. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  669.  
  670. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  671.  
  672. Max-Forwards: 70
  673.  
  674. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  675.  
  676. To: <sip:14162352999@216.58.0.51>
  677.  
  678. Contact: <sip:6477232824@99.227.42.4>
  679.  
  680. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  681.  
  682. CSeq: 102 INVITE
  683.  
  684. User-Agent: FPBX-2.9.0(1.6.2.15)
  685.  
  686. Date: Mon, 29 Aug 2011 17:09:06 GMT
  687.  
  688. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  689.  
  690. Supported: replaces, timer
  691.  
  692. Content-Type: application/sdp
  693.  
  694. Content-Length: 233
  695.  
  696.  
  697.  
  698. v=0
  699.  
  700. o=root 660440603 660440603 IN IP4 99.227.42.4
  701.  
  702. s=Asterisk PBX 1.6.2.15
  703.  
  704. c=IN IP4 99.227.42.4
  705.  
  706. t=0 0
  707.  
  708. m=audio 20004 RTP/AVP 0 101
  709.  
  710. a=rtpmap:0 PCMU/8000
  711.  
  712. a=rtpmap:101 telephone-event/8000
  713.  
  714. a=fmtp:101 0-16
  715.  
  716. a=ptime:20
  717.  
  718. a=sendrecv
  719.  
  720.  
  721. ---
  722.  
  723. pbx*CLI>
  724. Retransmitting #4 (no NAT) to 10.10.10.2:5060:
  725. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  726.  
  727. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  728.  
  729. Max-Forwards: 70
  730.  
  731. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  732.  
  733. To: <sip:14162352999@216.58.0.51>
  734.  
  735. Contact: <sip:6477232824@99.227.42.4>
  736.  
  737. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  738.  
  739. CSeq: 102 INVITE
  740.  
  741. User-Agent: FPBX-2.9.0(1.6.2.15)
  742.  
  743. Date: Mon, 29 Aug 2011 17:09:06 GMT
  744.  
  745. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  746.  
  747. Supported: replaces, timer
  748.  
  749. Content-Type: application/sdp
  750.  
  751. Content-Length: 233
  752.  
  753.  
  754.  
  755. v=0
  756.  
  757. o=root 660440603 660440603 IN IP4 99.227.42.4
  758.  
  759. s=Asterisk PBX 1.6.2.15
  760.  
  761. c=IN IP4 99.227.42.4
  762.  
  763. t=0 0
  764.  
  765. m=audio 20004 RTP/AVP 0 101
  766.  
  767. a=rtpmap:0 PCMU/8000
  768.  
  769. a=rtpmap:101 telephone-event/8000
  770.  
  771. a=fmtp:101 0-16
  772.  
  773. a=ptime:20
  774.  
  775. a=sendrecv
  776.  
  777.  
  778. ---
  779.  
  780. pbx*CLI>
  781. Retransmitting #5 (no NAT) to 10.10.10.2:5060:
  782. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  783.  
  784. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  785.  
  786. Max-Forwards: 70
  787.  
  788. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  789.  
  790. To: <sip:14162352999@216.58.0.51>
  791.  
  792. Contact: <sip:6477232824@99.227.42.4>
  793.  
  794. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  795.  
  796. CSeq: 102 INVITE
  797.  
  798. User-Agent: FPBX-2.9.0(1.6.2.15)
  799.  
  800. Date: Mon, 29 Aug 2011 17:09:06 GMT
  801.  
  802. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  803.  
  804. Supported: replaces, timer
  805.  
  806. Content-Type: application/sdp
  807.  
  808. Content-Length: 233
  809.  
  810.  
  811.  
  812. v=0
  813.  
  814. o=root 660440603 660440603 IN IP4 99.227.42.4
  815.  
  816. s=Asterisk PBX 1.6.2.15
  817.  
  818. c=IN IP4 99.227.42.4
  819.  
  820. t=0 0
  821.  
  822. m=audio 20004 RTP/AVP 0 101
  823.  
  824. a=rtpmap:0 PCMU/8000
  825.  
  826. a=rtpmap:101 telephone-event/8000
  827.  
  828. a=fmtp:101 0-16
  829.  
  830. a=ptime:20
  831.  
  832. a=sendrecv
  833.  
  834.  
  835. ---
  836.  
  837. pbx*CLI>
  838. Reliably Transmitting (no NAT) to 216.58.0.51:5060:
  839. OPTIONS sip:216.58.0.51 SIP/2.0
  840.  
  841. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK53ffc9c6;rport
  842.  
  843. Max-Forwards: 70
  844.  
  845. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as7197631f
  846.  
  847. To: <sip:216.58.0.51>
  848.  
  849. Contact: <sip:Unknown@99.227.42.4>
  850.  
  851. Call-ID: 6b2a6edf30e62bc6488b399135b06b52@99.227.42.4
  852.  
  853. CSeq: 102 OPTIONS
  854.  
  855. User-Agent: FPBX-2.9.0(1.6.2.15)
  856.  
  857. Date: Mon, 29 Aug 2011 17:09:11 GMT
  858.  
  859. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  860.  
  861. Supported: replaces, timer
  862.  
  863. Content-Length: 0
  864.  
  865.  
  866.  
  867.  
  868. ---
  869.  
  870. pbx*CLI>
  871. 
  872. <--- SIP read from UDP:216.58.0.51:5060 --->
  873. SIP/2.0 404 Not Found
  874.  
  875. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK53ffc9c6;received=99.227.42.4;rport=5060
  876.  
  877. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as7197631f
  878.  
  879. To: <sip:216.58.0.51>;tag=as3a327c0e
  880.  
  881. Call-ID: 6b2a6edf30e62bc6488b399135b06b52@99.227.42.4
  882.  
  883. CSeq: 102 OPTIONS
  884.  
  885. User-Agent: CIA.com PBX
  886.  
  887. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  888.  
  889. Supported: replaces
  890.  
  891. Accept: application/sdp
  892.  
  893. Content-Length: 0
  894.  
  895.  
  896.  
  897.  
  898. <------------->
  899.  
  900. pbx*CLI>
  901. --- (11 headers 0 lines) ---
  902.  
  903. pbx*CLI>
  904. Really destroying SIP dialog '6b2a6edf30e62bc6488b399135b06b52@99.227.42.4' Method: OPTIONS
  905.  
  906. pbx*CLI>
  907. Retransmitting #6 (no NAT) to 10.10.10.2:5060:
  908. INVITE sip:14162352999@216.58.0.51 SIP/2.0
  909.  
  910. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK63e8c96c;rport
  911.  
  912. Max-Forwards: 70
  913.  
  914. From: "6477232824" <sip:6477232824@cia.com>;tag=as06f9c85b
  915.  
  916. To: <sip:14162352999@216.58.0.51>
  917.  
  918. Contact: <sip:6477232824@99.227.42.4>
  919.  
  920. Call-ID: 5b72537b53c479b542af261f7ea2b1f0@cia.com
  921.  
  922. CSeq: 102 INVITE
  923.  
  924. User-Agent: FPBX-2.9.0(1.6.2.15)
  925.  
  926. Date: Mon, 29 Aug 2011 17:09:06 GMT
  927.  
  928. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  929.  
  930. Supported: replaces, timer
  931.  
  932. Content-Type: application/sdp
  933.  
  934. Content-Length: 233
  935.  
  936.  
  937.  
  938. v=0
  939.  
  940. o=root 660440603 660440603 IN IP4 99.227.42.4
  941.  
  942. s=Asterisk PBX 1.6.2.15
  943.  
  944. c=IN IP4 99.227.42.4
  945.  
  946. t=0 0
  947.  
  948. m=audio 20004 RTP/AVP 0 101
  949.  
  950. a=rtpmap:0 PCMU/8000
  951.  
  952. a=rtpmap:101 telephone-event/8000
  953.  
  954. a=fmtp:101 0-16
  955.  
  956. a=ptime:20
  957.  
  958. a=sendrecv
  959.  
  960.  
  961. ---
  962.  
  963. pbx*CLI>
  964.  == Manager 'admin' logged on from 127.0.0.1
  965.  
  966. pbx*CLI>
  967.  == Manager 'admin' logged off from 127.0.0.1
  968.  
  969. pbx*CLI>
  970. Reliably Transmitting (NAT) to 10.10.10.5:5061:
  971. OPTIONS sip:202@10.10.10.5:5061 SIP/2.0
  972.  
  973. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK5cc84dd2;rport
  974.  
  975. Max-Forwards: 70
  976.  
  977. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as1e239129
  978.  
  979. To: <sip:202@10.10.10.5:5061>
  980.  
  981. Contact: <sip:Unknown@10.10.10.7>
  982.  
  983. Call-ID: 34f92e2722348dde489eef004fb5860e@10.10.10.7
  984.  
  985. CSeq: 102 OPTIONS
  986.  
  987. User-Agent: FPBX-2.9.0(1.6.2.15)
  988.  
  989. Date: Mon, 29 Aug 2011 17:09:18 GMT
  990.  
  991. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  992.  
  993. Supported: replaces, timer
  994.  
  995. Content-Length: 0
  996.  
  997.  
  998.  
  999.  
  1000. ---
  1001.  
  1002. pbx*CLI>
  1003. 
  1004. <--- SIP read from UDP:10.10.10.5:5061 --->
  1005. SIP/2.0 200 OK
  1006.  
  1007. To: <sip:202@10.10.10.5:5061>;tag=81405397cfbf9f7ei1
  1008.  
  1009. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as1e239129
  1010.  
  1011. Call-ID: 34f92e2722348dde489eef004fb5860e@10.10.10.7
  1012.  
  1013. CSeq: 102 OPTIONS
  1014.  
  1015. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK5cc84dd2
  1016.  
  1017. Server: Linksys/SPA2102-3.3.6
  1018.  
  1019. Content-Length: 0
  1020.  
  1021. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  1022.  
  1023. Supported: x-sipura
  1024.  
  1025.  
  1026.  
  1027.  
  1028. <------------->
  1029.  
  1030. pbx*CLI>
  1031. --- (10 headers 0 lines) ---
  1032.  
  1033. pbx*CLI>
  1034. Really destroying SIP dialog '34f92e2722348dde489eef004fb5860e@10.10.10.7' Method: OPTIONS
  1035.  
  1036. pbx*CLI>
  1037.  == Everyone is busy/congested at this time (1:0/0/1)
  1038. -- Executing [s@macro-dialout-trunk:21] NoOp("SIP/203-00000014", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack
  1039. -- Executing [s@macro-dialout-trunk:22] Goto("SIP/203-00000014", "s-CHANUNAVAIL,1") in new stack
  1040. -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
  1041. -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/203-00000014", "RC=111") in new stack
  1042. -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/203-00000014", "111,1") in new stack
  1043.  
  1044. pbx*CLI>
  1045.  -- Goto (macro-dialout-trunk,111,1)
  1046. -- Executing [111@macro-dialout-trunk:1] Goto("SIP/203-00000014", "continue,1") in new stack
  1047.  
  1048. pbx*CLI>
  1049.  -- Goto (macro-dialout-trunk,continue,1)
  1050.  
  1051. pbx*CLI>
  1052.  -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/203-00000014", "1?noreport") in new stack
  1053. -- Goto (macro-dialout-trunk,continue,3)
  1054.  
  1055. pbx*CLI>
  1056.  -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/203-00000014", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 111 - failing through to other trunks") in new stack
  1057. -- Executing [continue@macro-dialout-trunk:4] Set("SIP/203-00000014", "CALLERID(number)=203") in new stack
  1058.  
  1059. pbx*CLI>
  1060.  -- Executing [4162352999@from-internal:6] Macro("SIP/203-00000014", "outisbusy,") in new stack
  1061.  
  1062. pbx*CLI>
  1063.  -- Executing [s@macro-outisbusy:1] Progress("SIP/203-00000014", "") in new stack
  1064.  
  1065. pbx*CLI>
  1066. Audio is at 10.10.10.7 port 20002
  1067. Adding codec 0x4 (ulaw) to SDP
  1068.  
  1069. pbx*CLI>
  1070. Adding non-codec 0x1 (telephone-event) to SDP
  1071.  
  1072. pbx*CLI>
  1073. 
  1074. <--- Transmitting (NAT) to 10.10.10.12:5060 --->
  1075. SIP/2.0 183 Session Progress
  1076.  
  1077. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a;received=10.10.10.12
  1078.  
  1079. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  1080.  
  1081. To: <sip:4162352999@10.10.10.7>;tag=as7c063a96
  1082.  
  1083. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  1084.  
  1085. CSeq: 102 INVITE
  1086.  
  1087. Server: FPBX-2.9.0(1.6.2.15)
  1088.  
  1089. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1090.  
  1091. Supported: replaces, timer
  1092.  
  1093. Contact: <sip:4162352999@10.10.10.7>
  1094.  
  1095. Content-Type: application/sdp
  1096.  
  1097. Content-Length: 233
  1098.  
  1099.  
  1100.  
  1101. v=0
  1102.  
  1103. o=root 1412649867 1412649867 IN IP4 10.10.10.7
  1104.  
  1105. s=Asterisk PBX 1.6.2.15
  1106.  
  1107. c=IN IP4 10.10.10.7
  1108.  
  1109. t=0 0
  1110.  
  1111. m=audio 20002 RTP/AVP 0 101
  1112.  
  1113. a=rtpmap:0 PCMU/8000
  1114.  
  1115. a=rtpmap:101 telephone-event/8000
  1116.  
  1117. a=fmtp:101 0-16
  1118.  
  1119. a=ptime:20
  1120.  
  1121. a=sendrecv
  1122.  
  1123.  
  1124. <------------>
  1125. -- Executing [s@macro-outisbusy:2] Playback("SIP/203-00000014", "all-circuits-busy-now,noanswer") in new stack
  1126.  
  1127. pbx*CLI>
  1128.  -- <SIP/203-00000014> Playing 'all-circuits-busy-now.gsm' (language 'en')
  1129.  
  1130. pbx*CLI>
  1131. Really destroying SIP dialog '5b72537b53c479b542af261f7ea2b1f0@cia.com' Method: INVITE
  1132.  
  1133. pbx*CLI>
  1134.  == Manager 'admin' logged on from 127.0.0.1
  1135.  
  1136. pbx*CLI>
  1137.  -- Executing [s@macro-outisbusy:3] Playback("SIP/203-00000014", "pls-try-call-later,noanswer") in new stack
  1138.  
  1139. pbx*CLI>
  1140.  -- <SIP/203-00000014> Playing 'pls-try-call-later.gsm' (language 'en')
  1141.  
  1142. pbx*CLI>
  1143. 
  1144. <--- SIP read from UDP:10.10.10.12:5060 --->
  1145. CANCEL sip:4162352999@10.10.10.7 SIP/2.0
  1146.  
  1147. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a
  1148.  
  1149. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  1150.  
  1151. To: <sip:4162352999@10.10.10.7>
  1152.  
  1153. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  1154.  
  1155. CSeq: 102 CANCEL
  1156.  
  1157. Max-Forwards: 70
  1158.  
  1159. Authorization: Digest username="203",realm="asterisk",nonce="7f6212ab",uri="sip:4162352999@10.10.10.7",algorithm=MD5,response="b0060eab0232a2c416158946df28f909"
  1160.  
  1161. User-Agent: Linksys/SPA942-6.1.5(a)
  1162.  
  1163. Content-Length: 0
  1164.  
  1165.  
  1166.  
  1167.  
  1168. <------------->
  1169. --- (10 headers 0 lines) ---
  1170. Sending to 10.10.10.12 : 5060 (NAT)
  1171.  
  1172. <--- Reliably Transmitting (NAT) to 10.10.10.12:5060 --->
  1173. SIP/2.0 487 Request Terminated
  1174.  
  1175. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a;received=10.10.10.12
  1176.  
  1177. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  1178.  
  1179. To: <sip:4162352999@10.10.10.7>;tag=as7c063a96
  1180.  
  1181. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  1182.  
  1183. CSeq: 102 INVITE
  1184.  
  1185. Server: FPBX-2.9.0(1.6.2.15)
  1186.  
  1187. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1188.  
  1189. Supported: replaces, timer
  1190.  
  1191. Content-Length: 0
  1192.  
  1193.  
  1194.  
  1195.  
  1196. <------------>
  1197.  
  1198. pbx*CLI>
  1199. 
  1200. <--- Transmitting (NAT) to 10.10.10.12:5060 --->
  1201. SIP/2.0 200 OK
  1202.  
  1203. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a;received=10.10.10.12
  1204.  
  1205. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  1206.  
  1207. To: <sip:4162352999@10.10.10.7>;tag=as7c063a96
  1208.  
  1209. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  1210.  
  1211. CSeq: 102 CANCEL
  1212.  
  1213. Server: FPBX-2.9.0(1.6.2.15)
  1214.  
  1215. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1216.  
  1217. Supported: replaces, timer
  1218.  
  1219. Content-Length: 0
  1220.  
  1221.  
  1222.  
  1223.  
  1224. <------------>
  1225.  
  1226. pbx*CLI>
  1227.  == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/203-00000014' in macro 'outisbusy'
  1228.  
  1229. pbx*CLI>
  1230.  == Spawn extension (from-internal, 4162352999, 6) exited non-zero on 'SIP/203-00000014'
  1231.  
  1232. pbx*CLI>
  1233.  -- Executing [h@from-internal:1] Hangup("SIP/203-00000014", "") in new stack
  1234. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-00000014'
  1235.  
  1236. pbx*CLI>
  1237. 
  1238. <--- SIP read from UDP:10.10.10.12:5060 --->
  1239. ACK sip:4162352999@10.10.10.7 SIP/2.0
  1240.  
  1241. Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-b577882a
  1242.  
  1243. From: "Itamar Desk" <sip:203@10.10.10.7>;tag=faaeaff47664e44ao0
  1244.  
  1245. To: <sip:4162352999@10.10.10.7>;tag=as7c063a96
  1246.  
  1247. Call-ID: 85ec10bc-d355ba22@10.10.10.12
  1248.  
  1249. CSeq: 102 ACK
  1250.  
  1251. Max-Forwards: 70
  1252.  
  1253. Authorization: Digest username="203",realm="asterisk",nonce="7f6212ab",uri="sip:4162352999@10.10.10.7",algorithm=MD5,response="20ca43714dd1f9007c4b7f6077299b2d"
  1254.  
  1255. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  1256.  
  1257. User-Agent: Linksys/SPA942-6.1.5(a)
  1258.  
  1259. Content-Length: 0
  1260.  
  1261.  
  1262.  
  1263.  
  1264. <------------->
  1265. --- (11 headers 0 lines) ---
  1266. Really destroying SIP dialog '85ec10bc-d355ba22@10.10.10.12' Method: ACK
  1267.  
  1268. pbx*CLI>
  1269.  == Manager 'admin' logged off from 127.0.0.1
  1270.  
  1271. pbx*CLI>
  1272.  == Manager 'admin' logged on from 127.0.0.1
  1273.  
  1274. pbx*CLI>
  1275.  == Manager 'admin' logged off from 127.0.0.1
  1276.  
  1277. pbx*CLI> exit
  1278.  
  1279. Executing last minute cleanups
  1280. ]0;root@pbx:~[root@pbx ~]# exit
  1281. logout
  1282. 
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