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  1. =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.08.28 20:56:00 =~=~=~=~=~=~=~=~=~=~=~=
  2. login as: root
  3. root@10.10.10.7's password:
  4. Last login: Sun Aug 28 19:45:33 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. Core debug is at least 10
  19.  
  20. pbx*CLI>
  21. Reliably Transmitting (no NAT) to 216.58.0.51:5060:
  22. OPTIONS sip:216.58.0.51 SIP/2.0
  23.  
  24. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK78adc394;rport
  25.  
  26. Max-Forwards: 70
  27.  
  28. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as5868e072
  29.  
  30. To: <sip:216.58.0.51>
  31.  
  32. Contact: <sip:Unknown@99.227.42.4>
  33.  
  34. Call-ID: 680be2045073f3600f4c312f353b9f7a@99.227.42.4
  35.  
  36. CSeq: 102 OPTIONS
  37.  
  38. User-Agent: FPBX-2.9.0(1.6.2.15)
  39.  
  40. Date: Mon, 29 Aug 2011 00:37:45 GMT
  41.  
  42. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  43.  
  44. Supported: replaces, timer
  45.  
  46. Content-Length: 0
  47.  
  48.  
  49.  
  50.  
  51. ---
  52.  
  53. pbx*CLI>
  54. 
  55. <--- SIP read from UDP:216.58.0.51:5060 --->
  56. SIP/2.0 404 Not Found
  57.  
  58. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK78adc394;received=99.227.42.4;rport=5060
  59.  
  60. From: "Unknown" <sip:Unknown@99.227.42.4>;tag=as5868e072
  61.  
  62. To: <sip:216.58.0.51>;tag=as2449abd1
  63.  
  64. Call-ID: 680be2045073f3600f4c312f353b9f7a@99.227.42.4
  65.  
  66. CSeq: 102 OPTIONS
  67.  
  68. User-Agent: CIA.com PBX
  69.  
  70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  71.  
  72. Supported: replaces
  73.  
  74. Accept: application/sdp
  75.  
  76. Content-Length: 0
  77.  
  78.  
  79.  
  80.  
  81. <------------->
  82. --- (11 headers 0 lines) ---
  83. Really destroying SIP dialog '680be2045073f3600f4c312f353b9f7a@99.227.42.4' Method: OPTIONS
  84.  
  85. pbx*CLI>
  86. Reliably Transmitting (NAT) to 10.10.10.5:5061:
  87. OPTIONS sip:202@10.10.10.5:5061 SIP/2.0
  88.  
  89. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6ca4d8a6;rport
  90.  
  91. Max-Forwards: 70
  92.  
  93. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as5ca8827e
  94.  
  95. To: <sip:202@10.10.10.5:5061>
  96.  
  97. Contact: <sip:Unknown@10.10.10.7>
  98.  
  99. Call-ID: 33e1fd576c8c575a1fe7c6505fb75e0d@10.10.10.7
  100.  
  101. CSeq: 102 OPTIONS
  102.  
  103. User-Agent: FPBX-2.9.0(1.6.2.15)
  104.  
  105. Date: Mon, 29 Aug 2011 00:37:53 GMT
  106.  
  107. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  108.  
  109. Supported: replaces, timer
  110.  
  111. Content-Length: 0
  112.  
  113.  
  114.  
  115.  
  116. ---
  117.  
  118. pbx*CLI>
  119. 
  120. <--- SIP read from UDP:10.10.10.5:5061 --->
  121. SIP/2.0 200 OK
  122.  
  123. To: <sip:202@10.10.10.5:5061>;tag=81405397cfbf9f7ei1
  124.  
  125. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as5ca8827e
  126.  
  127. Call-ID: 33e1fd576c8c575a1fe7c6505fb75e0d@10.10.10.7
  128.  
  129. CSeq: 102 OPTIONS
  130.  
  131. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6ca4d8a6
  132.  
  133. Server: Linksys/SPA2102-3.3.6
  134.  
  135. Content-Length: 0
  136.  
  137. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  138.  
  139. Supported: x-sipura
  140.  
  141.  
  142.  
  143.  
  144. <------------->
  145. --- (10 headers 0 lines) ---
  146. Really destroying SIP dialog '33e1fd576c8c575a1fe7c6505fb75e0d@10.10.10.7' Method: OPTIONS
  147.  
  148. pbx*CLI>
  149. 
  150. <--- SIP read from UDP:216.58.0.51:5060 --->
  151. INVITE sip:6477232824@99.227.42.4 SIP/2.0
  152.  
  153. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK6b2910f5;rport
  154.  
  155. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  156.  
  157. To: <sip:6477232824@99.227.42.4>
  158.  
  159. Contact: <sip:4168853558@216.58.0.51>
  160.  
  161. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  162.  
  163. CSeq: 102 INVITE
  164.  
  165. User-Agent: CIA.com PBX
  166.  
  167. Max-Forwards: 70
  168.  
  169. Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
  170.  
  171. Date: Mon, 29 Aug 2011 00:56:33 GMT
  172.  
  173. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  174.  
  175. Supported: replaces
  176.  
  177. Content-Type: application/sdp
  178.  
  179. Content-Length: 285
  180.  
  181.  
  182.  
  183. v=0
  184.  
  185. o=root 20246 20246 IN IP4 216.58.0.51
  186.  
  187. s=session
  188.  
  189. c=IN IP4 216.58.0.51
  190.  
  191. t=0 0
  192.  
  193. m=audio 11990 RTP/AVP 0 8 3 101
  194.  
  195. a=rtpmap:0 PCMU/8000
  196.  
  197. a=rtpmap:8 PCMA/8000
  198.  
  199. a=rtpmap:3 GSM/8000
  200.  
  201. a=rtpmap:101 telephone-event/8000
  202.  
  203. a=fmtp:101 0-16
  204.  
  205. a=silenceSupp:off - - - -
  206.  
  207. a=ptime:20
  208.  
  209. a=sendrecv
  210.  
  211.  
  212. <------------->
  213.  
  214. pbx*CLI>
  215. --- (15 headers 14 lines) ---
  216.  
  217. pbx*CLI>
  218.  == Using SIP RTP TOS bits 184
  219.  
  220. pbx*CLI>
  221.  == Using SIP RTP CoS mark 5
  222.  
  223. pbx*CLI>
  224. Sending to 216.58.0.51 : 5060 (NAT)
  225.  
  226. pbx*CLI>
  227. Using INVITE request as basis request - 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  228.  
  229. pbx*CLI>
  230. Found peer '6477232824' for '4168853558' from 216.58.0.51:5060
  231.  
  232. pbx*CLI>
  233. Found RTP audio format 0
  234.  
  235. pbx*CLI>
  236. Found RTP audio format 8
  237. Found RTP audio format 3
  238. Found RTP audio format 101
  239.  
  240. pbx*CLI>
  241. Found audio description format PCMU for ID 0
  242.  
  243. pbx*CLI>
  244. Found audio description format PCMA for ID 8
  245.  
  246. pbx*CLI>
  247. Found audio description format GSM for ID 3
  248.  
  249. pbx*CLI>
  250. Found audio description format telephone-event for ID 101
  251.  
  252. pbx*CLI>
  253. Capabilities: us - 0x4 (ulaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  254.  
  255. pbx*CLI>
  256. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  257.  
  258. pbx*CLI>
  259. Peer audio RTP is at port 216.58.0.51:11990
  260.  
  261. pbx*CLI>
  262. Looking for 6477232824 in from-trunk (domain 99.227.42.4)
  263.  
  264. pbx*CLI>
  265. list_route: hop: <sip:4168853558@216.58.0.51>
  266.  
  267. pbx*CLI>
  268. 
  269. <--- Transmitting (no NAT) to 216.58.0.51:5060 --->
  270. SIP/2.0 100 Trying
  271.  
  272. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK6b2910f5;received=216.58.0.51;rport=5060
  273.  
  274. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  275.  
  276. To: <sip:6477232824@99.227.42.4>
  277.  
  278. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  279.  
  280. CSeq: 102 INVITE
  281.  
  282. Server: FPBX-2.9.0(1.6.2.15)
  283.  
  284. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  285.  
  286. Supported: replaces, timer
  287.  
  288. Contact: <sip:6477232824@99.227.42.4>
  289.  
  290. Content-Length: 0
  291.  
  292.  
  293.  
  294.  
  295. <------------>
  296.  
  297. pbx*CLI>
  298.  -- Executing [6477232824@from-trunk:1] Set("SIP/6477232824-0000003e", "__FROM_DID=6477232824") in new stack
  299.  
  300. pbx*CLI>
  301.  -- Executing [6477232824@from-trunk:2] Gosub("SIP/6477232824-0000003e", "app-blacklist-check,s,1") in new stack
  302.  
  303. pbx*CLI>
  304.  -- Executing [s@app-blacklist-check:1] GotoIf("SIP/6477232824-0000003e", "0?blacklisted") in new stack
  305.  
  306. pbx*CLI>
  307.  -- Executing [s@app-blacklist-check:2] Set("SIP/6477232824-0000003e", "CALLED_BLACKLIST=1") in new stack
  308.  
  309. pbx*CLI>
  310.  -- Executing [s@app-blacklist-check:3] Return("SIP/6477232824-0000003e", "") in new stack
  311.  
  312. pbx*CLI>
  313.  -- Executing [6477232824@from-trunk:3] ExecIf("SIP/6477232824-0000003e", "0 ?Set(CALLERID(name)=4168853558)") in new stack
  314. -- Executing [6477232824@from-trunk:4] Set("SIP/6477232824-0000003e", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  315.  
  316. pbx*CLI>
  317.  -- Executing [6477232824@from-trunk:5] Set("SIP/6477232824-0000003e", "CALLERPRES()=allowed_not_screened") in new stack
  318.  
  319. pbx*CLI>
  320.  -- Executing [6477232824@from-trunk:6] Goto("SIP/6477232824-0000003e", "ext-group,601,1") in new stack
  321. -- Goto (ext-group,601,1)
  322. -- Executing [601@ext-group:1] Macro("SIP/6477232824-0000003e", "user-callerid,") in new stack
  323.  
  324. pbx*CLI>
  325.  -- Executing [s@macro-user-callerid:1] Set("SIP/6477232824-0000003e", "AMPUSER=4168853558") in new stack
  326.  
  327. pbx*CLI>
  328.  -- Executing [s@macro-user-callerid:2] GotoIf("SIP/6477232824-0000003e", "0?report") in new stack
  329.  
  330. pbx*CLI>
  331.  -- Executing [s@macro-user-callerid:3] ExecIf("SIP/6477232824-0000003e", "1?Set(REALCALLERIDNUM=4168853558)") in new stack
  332.  
  333. pbx*CLI>
  334.  -- Executing [s@macro-user-callerid:4] Set("SIP/6477232824-0000003e", "AMPUSER=") in new stack
  335.  
  336. pbx*CLI>
  337.  -- Executing [s@macro-user-callerid:5] Set("SIP/6477232824-0000003e", "AMPUSERCIDNAME=") in new stack
  338.  
  339. pbx*CLI>
  340.  -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6477232824-0000003e", "1?report") in new stack
  341.  
  342. pbx*CLI>
  343.  -- Goto (macro-user-callerid,s,11)
  344.  
  345. pbx*CLI>
  346.  -- Executing [s@macro-user-callerid:11] GotoIf("SIP/6477232824-0000003e", "0?continue") in new stack
  347.  
  348. pbx*CLI>
  349.  -- Executing [s@macro-user-callerid:12] Set("SIP/6477232824-0000003e", "__TTL=64") in new stack
  350.  
  351. pbx*CLI>
  352.  -- Executing [s@macro-user-callerid:13] GotoIf("SIP/6477232824-0000003e", "1?continue") in new stack
  353.  
  354. pbx*CLI>
  355.  -- Goto (macro-user-callerid,s,24)
  356.  
  357. pbx*CLI>
  358.  -- Executing [s@macro-user-callerid:24] Set("SIP/6477232824-0000003e", "CALLERID(number)=4168853558") in new stack
  359.  
  360. pbx*CLI>
  361.  -- Executing [s@macro-user-callerid:25] Set("SIP/6477232824-0000003e", "CALLERID(name)=Unknown") in new stack
  362.  
  363. pbx*CLI>
  364.  -- Executing [601@ext-group:2] Macro("SIP/6477232824-0000003e", "blkvm-setifempty,") in new stack
  365.  
  366. pbx*CLI>
  367.  -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/6477232824-0000003e", "1?init") in new stack
  368.  
  369. pbx*CLI>
  370.  -- Goto (macro-blkvm-setifempty,s,4)
  371.  
  372. pbx*CLI>
  373.  -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/6477232824-0000003e", "__BLKVM_CHANNEL=SIP/6477232824-0000003e") in new stack
  374.  
  375. pbx*CLI>
  376.  -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/6477232824-0000003e", "SHARED(BLKVM,SIP/6477232824-0000003e)=TRUE") in new stack
  377.  
  378. pbx*CLI>
  379.  -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/6477232824-0000003e", "GOSUB_RETVAL=TRUE") in new stack
  380.  
  381. pbx*CLI>
  382.  -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/6477232824-0000003e", "") in new stack
  383.  
  384. pbx*CLI>
  385.  -- Executing [601@ext-group:3] GotoIf("SIP/6477232824-0000003e", "1?skipov") in new stack
  386.  
  387. pbx*CLI>
  388.  -- Goto (ext-group,601,6)
  389.  
  390. pbx*CLI>
  391.  -- Executing [601@ext-group:6] Set("SIP/6477232824-0000003e", "RRNODEST=") in new stack
  392.  
  393. pbx*CLI>
  394.  -- Executing [601@ext-group:7] Set("SIP/6477232824-0000003e", "__NODEST=601") in new stack
  395.  
  396. pbx*CLI>
  397.  -- Executing [601@ext-group:8] GosubIf("SIP/6477232824-0000003e", "0?sub-rgsetcid,s,1") in new stack
  398.  
  399. pbx*CLI>
  400.  -- Executing [601@ext-group:9] Set("SIP/6477232824-0000003e", "RecordMethod=Group") in new stack
  401.  
  402. pbx*CLI>
  403.  -- Executing [601@ext-group:10] Macro("SIP/6477232824-0000003e", "record-enable,201-203,Group") in new stack
  404.  
  405. pbx*CLI>
  406.  -- Executing [s@macro-record-enable:1] GotoIf("SIP/6477232824-0000003e", "1?check") in new stack
  407.  
  408. pbx*CLI>
  409.  -- Goto (macro-record-enable,s,4)
  410.  
  411. pbx*CLI>
  412.  -- Executing [s@macro-record-enable:4] ExecIf("SIP/6477232824-0000003e", "0?MacroExit()") in new stack
  413.  
  414. pbx*CLI>
  415.  -- Executing [s@macro-record-enable:5] GotoIf("SIP/6477232824-0000003e", "1?Group:OUT") in new stack
  416.  
  417. pbx*CLI>
  418.  -- Goto (macro-record-enable,s,6)
  419.  
  420. pbx*CLI>
  421.  -- Executing [s@macro-record-enable:6] Set("SIP/6477232824-0000003e", "LOOPCNT=2") in new stack
  422.  
  423. pbx*CLI>
  424.  -- Executing [s@macro-record-enable:7] Set("SIP/6477232824-0000003e", "ITER=1") in new stack
  425.  
  426. pbx*CLI>
  427.  -- Executing [s@macro-record-enable:8] GotoIf("SIP/6477232824-0000003e", "1?continue") in new stack
  428.  
  429. pbx*CLI>
  430.  -- Goto (macro-record-enable,s,12)
  431.  
  432. pbx*CLI>
  433.  -- Executing [s@macro-record-enable:12] Set("SIP/6477232824-0000003e", "ITER=2") in new stack
  434.  
  435. pbx*CLI>
  436.  -- Executing [s@macro-record-enable:13] GotoIf("SIP/6477232824-0000003e", "1?begin") in new stack
  437.  
  438. pbx*CLI>
  439.  -- Goto (macro-record-enable,s,8)
  440.  
  441. pbx*CLI>
  442.  -- Executing [s@macro-record-enable:8] GotoIf("SIP/6477232824-0000003e", "1?continue") in new stack
  443.  
  444. pbx*CLI>
  445.  -- Goto (macro-record-enable,s,12)
  446.  
  447. pbx*CLI>
  448.  -- Executing [s@macro-record-enable:12] Set("SIP/6477232824-0000003e", "ITER=3") in new stack
  449.  
  450. pbx*CLI>
  451.  -- Executing [s@macro-record-enable:13] GotoIf("SIP/6477232824-0000003e", "0?begin") in new stack
  452.  
  453. pbx*CLI>
  454.  -- Executing [s@macro-record-enable:14] GotoIf("SIP/6477232824-0000003e", "0?IN") in new stack
  455.  
  456. pbx*CLI>
  457.  -- Executing [s@macro-record-enable:15] ExecIf("SIP/6477232824-0000003e", "1?MacroExit()") in new stack
  458.  
  459. pbx*CLI>
  460.  -- Executing [601@ext-group:11] Set("SIP/6477232824-0000003e", "RingGroupMethod=ringall") in new stack
  461.  
  462. pbx*CLI>
  463.  -- Executing [601@ext-group:12] Macro("SIP/6477232824-0000003e", "dial,20,tr,201-203") in new stack
  464.  
  465. pbx*CLI>
  466.  -- Executing [s@macro-dial:1] GotoIf("SIP/6477232824-0000003e", "1?dial") in new stack
  467.  
  468. pbx*CLI>
  469.  -- Goto (macro-dial,s,3)
  470.  
  471. pbx*CLI>
  472.  -- Executing [s@macro-dial:3] AGI("SIP/6477232824-0000003e", "dialparties.agi") in new stack
  473.  
  474. pbx*CLI>
  475.  -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  476.  
  477. pbx*CLI>
  478.  dialparties.agi: Starting New Dialparties.agi
  479.  
  480. pbx*CLI>
  481.  dialparties.agi: Caller ID name is 'Unknown' number is '4168853558'
  482.  
  483. pbx*CLI>
  484.  > dialparties.agi: USE_CONFIRMATION: 'FALSE'
  485.  
  486. pbx*CLI>
  487.  > dialparties.agi: RINGGROUP_INDEX: ''
  488.  
  489. pbx*CLI>
  490.  dialparties.agi: Methodology of ring is 'ringall'
  491.  
  492. pbx*CLI>
  493.  -- dialparties.agi: Added extension 201 to extension map
  494.  
  495. pbx*CLI>
  496.  -- dialparties.agi: Added extension 203 to extension map
  497.  
  498. pbx*CLI>
  499.  -- dialparties.agi: Extension 201 cf is disabled
  500.  
  501. pbx*CLI>
  502.  -- dialparties.agi: Extension 203 cf is disabled
  503.  
  504. pbx*CLI>
  505.  -- dialparties.agi: Extension 201 do not disturb is disabled
  506.  
  507. pbx*CLI>
  508.  -- dialparties.agi: Extension 203 do not disturb is disabled
  509.  
  510. pbx*CLI>
  511.  > dialparties.agi: extnum 201 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  512.  
  513. pbx*CLI>
  514.  -- dialparties.agi: dbset CALLTRACE/201 to 4168853558
  515.  
  516. pbx*CLI>
  517.  > dialparties.agi: extnum 203 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  518.  
  519. pbx*CLI>
  520.  -- dialparties.agi: dbset CALLTRACE/203 to 4168853558
  521.  
  522. pbx*CLI>
  523.  -- dialparties.agi: Filtered ARG3: 201-203
  524.  
  525. pbx*CLI>
  526.  > dialparties.agi: NODEST: 601 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
  527.  
  528. pbx*CLI>
  529.  > dialparties.agi: NODEST: 601 blkvm enabled macro already in dialopts: trM(auto-blkvm)
  530.  
  531. pbx*CLI>
  532.  -- <SIP/6477232824-0000003e>AGI Script dialparties.agi completed, returning 0
  533.  
  534. pbx*CLI>
  535.  -- Executing [s@macro-dial:7] Dial("SIP/6477232824-0000003e", "SIP/201&SIP/203,20,trM(auto-blkvm)") in new stack
  536.  
  537. pbx*CLI>
  538.  == Using SIP RTP TOS bits 184
  539.  
  540. pbx*CLI>
  541.  == Using SIP RTP CoS mark 5
  542.  
  543. pbx*CLI>
  544. Really destroying SIP dialog '26a7879465be89d47353c9ad623ff31c@127.0.0.1' Method: INVITE
  545.  
  546. pbx*CLI>
  547.  == Using SIP RTP TOS bits 184
  548.  
  549. pbx*CLI>
  550.  == Using SIP RTP CoS mark 5
  551.  
  552. pbx*CLI>
  553. Audio is at 10.10.10.7 port 20002
  554.  
  555. pbx*CLI>
  556. Adding codec 0x4 (ulaw) to SDP
  557.  
  558. pbx*CLI>
  559. Adding non-codec 0x1 (telephone-event) to SDP
  560.  
  561. pbx*CLI>
  562. Reliably Transmitting (NAT) to 10.10.10.12:5060:
  563. INVITE sip:203@10.10.10.12:5060 SIP/2.0
  564.  
  565. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK0f44986e;rport
  566.  
  567. Max-Forwards: 70
  568.  
  569. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  570.  
  571. To: <sip:203@10.10.10.12:5060>
  572.  
  573. Contact: <sip:4168853558@10.10.10.7>
  574.  
  575. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  576.  
  577. CSeq: 102 INVITE
  578.  
  579. User-Agent: FPBX-2.9.0(1.6.2.15)
  580.  
  581. Date: Mon, 29 Aug 2011 00:37:54 GMT
  582.  
  583. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  584.  
  585. Supported: replaces, timer
  586.  
  587. Content-Type: application/sdp
  588.  
  589. Content-Length: 233
  590.  
  591.  
  592.  
  593. v=0
  594.  
  595. o=root 1041144182 1041144182 IN IP4 10.10.10.7
  596.  
  597. s=Asterisk PBX 1.6.2.15
  598.  
  599. c=IN IP4 10.10.10.7
  600.  
  601. t=0 0
  602.  
  603. m=audio 20002 RTP/AVP 0 101
  604.  
  605. a=rtpmap:0 PCMU/8000
  606.  
  607. a=rtpmap:101 telephone-event/8000
  608.  
  609. a=fmtp:101 0-16
  610.  
  611. a=ptime:20
  612.  
  613. a=sendrecv
  614.  
  615.  
  616. ---
  617. -- Called 203
  618.  
  619. pbx*CLI>
  620. 
  621. <--- Transmitting (no NAT) to 216.58.0.51:5060 --->
  622. SIP/2.0 180 Ringing
  623.  
  624. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK6b2910f5;received=216.58.0.51;rport=5060
  625.  
  626. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  627.  
  628. To: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  629.  
  630. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  631.  
  632. CSeq: 102 INVITE
  633.  
  634. Server: FPBX-2.9.0(1.6.2.15)
  635.  
  636. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  637.  
  638. Supported: replaces, timer
  639.  
  640. Contact: <sip:6477232824@99.227.42.4>
  641.  
  642. Content-Length: 0
  643.  
  644.  
  645.  
  646.  
  647. <------------>
  648.  
  649. pbx*CLI>
  650. 
  651. <--- SIP read from UDP:10.10.10.12:5060 --->
  652. SIP/2.0 100 Trying
  653.  
  654. To: <sip:203@10.10.10.12:5060>
  655.  
  656. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  657.  
  658. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  659.  
  660. CSeq: 102 INVITE
  661.  
  662. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK0f44986e
  663.  
  664. Server: Linksys/SPA942-6.1.5(a)
  665.  
  666. Content-Length: 0
  667.  
  668.  
  669.  
  670.  
  671. <------------->
  672.  
  673. pbx*CLI>
  674. --- (8 headers 0 lines) ---
  675.  
  676. pbx*CLI>
  677. 
  678. <--- SIP read from UDP:10.10.10.12:5060 --->
  679. SIP/2.0 180 Ringing
  680.  
  681. To: <sip:203@10.10.10.12:5060>;tag=e52b3a3cb5e9dd53i0
  682.  
  683. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  684.  
  685. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  686.  
  687. CSeq: 102 INVITE
  688.  
  689. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK0f44986e
  690.  
  691. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  692.  
  693. Server: Linksys/SPA942-6.1.5(a)
  694.  
  695. Content-Length: 0
  696.  
  697.  
  698.  
  699.  
  700. <------------->
  701.  
  702. pbx*CLI>
  703. --- (9 headers 0 lines) ---
  704.  
  705. pbx*CLI>
  706.  -- SIP/203-0000003f is ringing
  707.  
  708. pbx*CLI>
  709. 
  710. <--- Transmitting (no NAT) to 216.58.0.51:5060 --->
  711. SIP/2.0 180 Ringing
  712.  
  713. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK6b2910f5;received=216.58.0.51;rport=5060
  714.  
  715. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  716.  
  717. To: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  718.  
  719. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  720.  
  721. CSeq: 102 INVITE
  722.  
  723. Server: FPBX-2.9.0(1.6.2.15)
  724.  
  725. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  726.  
  727. Supported: replaces, timer
  728.  
  729. Contact: <sip:6477232824@99.227.42.4>
  730.  
  731. Content-Length: 0
  732.  
  733.  
  734.  
  735.  
  736. <------------>
  737.  
  738. pbx*CLI>
  739. 
  740. <--- SIP read from UDP:10.10.10.12:5060 --->
  741. SIP/2.0 200 OK
  742.  
  743. To: <sip:203@10.10.10.12:5060>;tag=e52b3a3cb5e9dd53i0
  744.  
  745. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  746.  
  747. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  748.  
  749. CSeq: 102 INVITE
  750.  
  751. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK0f44986e
  752.  
  753. Contact: "Itamar Desk" <sip:203@10.10.10.12:5060>
  754.  
  755. Server: Linksys/SPA942-6.1.5(a)
  756.  
  757. Content-Length: 208
  758.  
  759. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  760.  
  761. Supported: replaces
  762.  
  763. Content-Type: application/sdp
  764.  
  765.  
  766.  
  767. v=0
  768.  
  769. o=- 15663072 15663072 IN IP4 10.10.10.12
  770.  
  771. s=-
  772.  
  773. c=IN IP4 10.10.10.12
  774.  
  775. t=0 0
  776.  
  777. m=audio 13513 RTP/AVP 0 101
  778.  
  779. a=rtpmap:0 PCMU/8000
  780.  
  781. a=rtpmap:101 telephone-event/8000
  782.  
  783. a=fmtp:101 0-15
  784.  
  785. a=ptime:30
  786.  
  787. a=sendrecv
  788.  
  789.  
  790. <------------->
  791.  
  792. pbx*CLI>
  793. --- (12 headers 11 lines) ---
  794.  
  795. pbx*CLI>
  796. Found RTP audio format 0
  797. Found RTP audio format 101
  798. Found audio description format PCMU for ID 0
  799.  
  800. pbx*CLI>
  801. Found audio description format telephone-event for ID 101
  802.  
  803. pbx*CLI>
  804. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  805.  
  806. pbx*CLI>
  807. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  808.  
  809. pbx*CLI>
  810. Peer audio RTP is at port 10.10.10.12:13513
  811.  
  812. pbx*CLI>
  813. list_route: hop: <sip:203@10.10.10.12:5060>
  814.  
  815. pbx*CLI>
  816. set_destination: Parsing <sip:203@10.10.10.12:5060> for address/port to send to
  817. set_destination: set destination to 10.10.10.12, port 5060
  818.  
  819. pbx*CLI>
  820. Transmitting (NAT) to 10.10.10.12:5060:
  821. ACK sip:203@10.10.10.12:5060 SIP/2.0
  822.  
  823. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK0f7b58d2;rport
  824.  
  825. Max-Forwards: 70
  826.  
  827. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  828.  
  829. To: <sip:203@10.10.10.12:5060>;tag=e52b3a3cb5e9dd53i0
  830.  
  831. Contact: <sip:4168853558@10.10.10.7>
  832.  
  833. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  834.  
  835. CSeq: 102 ACK
  836.  
  837. User-Agent: FPBX-2.9.0(1.6.2.15)
  838.  
  839. Content-Length: 0
  840.  
  841.  
  842.  
  843.  
  844. ---
  845.  
  846. pbx*CLI>
  847.  -- SIP/203-0000003f answered SIP/6477232824-0000003e
  848.  
  849. pbx*CLI>
  850.  -- Executing [s@macro-auto-blkvm:1] Set("SIP/203-0000003f", "__MACRO_RESULT=") in new stack
  851.  
  852. pbx*CLI>
  853.  -- Executing [s@macro-auto-blkvm:2] Macro("SIP/203-0000003f", "blkvm-clr,") in new stack
  854.  
  855. pbx*CLI>
  856.  -- Executing [s@macro-blkvm-clr:1] Set("SIP/203-0000003f", "SHARED(BLKVM,SIP/6477232824-0000003e)=") in new stack
  857.  
  858. pbx*CLI>
  859.  -- Executing [s@macro-blkvm-clr:2] Set("SIP/203-0000003f", "GOSUB_RETVAL=") in new stack
  860.  
  861. pbx*CLI>
  862.  -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/203-0000003f", "") in new stack
  863.  
  864. pbx*CLI>
  865. Audio is at 99.227.42.4 port 20004
  866.  
  867. pbx*CLI>
  868. Adding codec 0x4 (ulaw) to SDP
  869.  
  870. pbx*CLI>
  871. Adding non-codec 0x1 (telephone-event) to SDP
  872.  
  873. pbx*CLI>
  874. 
  875. <--- Reliably Transmitting (no NAT) to 216.58.0.51:5060 --->
  876. SIP/2.0 200 OK
  877.  
  878. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK6b2910f5;received=216.58.0.51;rport=5060
  879.  
  880. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  881.  
  882. To: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  883.  
  884. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  885.  
  886. CSeq: 102 INVITE
  887.  
  888. Server: FPBX-2.9.0(1.6.2.15)
  889.  
  890. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  891.  
  892. Supported: replaces, timer
  893.  
  894. Contact: <sip:6477232824@99.227.42.4>
  895.  
  896. Content-Type: application/sdp
  897.  
  898. Content-Length: 233
  899.  
  900.  
  901.  
  902. v=0
  903.  
  904. o=root 577007860 577007860 IN IP4 99.227.42.4
  905.  
  906. s=Asterisk PBX 1.6.2.15
  907.  
  908. c=IN IP4 99.227.42.4
  909.  
  910. t=0 0
  911.  
  912. m=audio 20004 RTP/AVP 0 101
  913.  
  914. a=rtpmap:0 PCMU/8000
  915.  
  916. a=rtpmap:101 telephone-event/8000
  917.  
  918. a=fmtp:101 0-16
  919.  
  920. a=ptime:20
  921.  
  922. a=sendrecv
  923.  
  924.  
  925. <------------>
  926.  
  927. pbx*CLI>
  928. 
  929. <--- SIP read from UDP:216.58.0.51:5060 --->
  930. ACK sip:6477232824@99.227.42.4 SIP/2.0
  931.  
  932. Via: SIP/2.0/UDP 216.58.0.51:5060;branch=z9hG4bK01878a16;rport
  933.  
  934. From: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  935.  
  936. To: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  937.  
  938. Contact: <sip:4168853558@216.58.0.51>
  939.  
  940. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  941.  
  942. CSeq: 102 ACK
  943.  
  944. User-Agent: CIA.com PBX
  945.  
  946. Max-Forwards: 70
  947.  
  948. Remote-Party-ID: "Unknown" <sip:4168853558@216.58.0.51>;privacy=off;screen=no
  949.  
  950. Content-Length: 0
  951.  
  952.  
  953. pbx*CLI>
  954. 
  955.  
  956.  
  957. <------------->
  958.  
  959. pbx*CLI>
  960. --- (11 headers 0 lines) ---
  961.  
  962. pbx*CLI>
  963. Really destroying SIP dialog '3a88375d6bf2eb3923db8ba3676e75f3@127.0.0.1' Method: REGISTER
  964.  
  965. pbx*CLI>
  966. Reliably Transmitting (NAT) to 10.10.10.12:5060:
  967. OPTIONS sip:203@10.10.10.12:5060 SIP/2.0
  968.  
  969. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK67c6a8fe;rport
  970.  
  971. Max-Forwards: 70
  972.  
  973. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as538c9fe9
  974.  
  975. To: <sip:203@10.10.10.12:5060>
  976.  
  977. Contact: <sip:Unknown@10.10.10.7>
  978.  
  979. Call-ID: 40d9f42f37fc28b457c0f98c063936d1@10.10.10.7
  980.  
  981. CSeq: 102 OPTIONS
  982.  
  983. User-Agent: FPBX-2.9.0(1.6.2.15)
  984.  
  985. Date: Mon, 29 Aug 2011 00:38:21 GMT
  986.  
  987. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  988.  
  989. Supported: replaces, timer
  990.  
  991. Content-Length: 0
  992.  
  993.  
  994.  
  995.  
  996. ---
  997.  
  998. pbx*CLI>
  999. 
  1000. <--- SIP read from UDP:10.10.10.12:5060 --->
  1001. SIP/2.0 200 OK
  1002.  
  1003. To: <sip:203@10.10.10.12:5060>;tag=cbd728af33a50f31i0
  1004.  
  1005. From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as538c9fe9
  1006.  
  1007. Call-ID: 40d9f42f37fc28b457c0f98c063936d1@10.10.10.7
  1008.  
  1009. CSeq: 102 OPTIONS
  1010.  
  1011. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK67c6a8fe
  1012.  
  1013. Server: Linksys/SPA942-6.1.5(a)
  1014.  
  1015. Content-Length: 0
  1016.  
  1017. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  1018.  
  1019. Supported: replaces
  1020.  
  1021.  
  1022.  
  1023.  
  1024. <------------->
  1025.  
  1026. pbx*CLI>
  1027. --- (10 headers 0 lines) ---
  1028.  
  1029. pbx*CLI>
  1030. Really destroying SIP dialog '40d9f42f37fc28b457c0f98c063936d1@10.10.10.7' Method: OPTIONS
  1031.  
  1032. pbx*CLI>
  1033.  -- Executing [h@macro-dial:1] Macro("SIP/6477232824-0000003e", "hangupcall") in new stack
  1034.  
  1035. pbx*CLI>
  1036.  -- Executing [s@macro-hangupcall:1] GotoIf("SIP/6477232824-0000003e", "1?theend") in new stack
  1037.  
  1038. pbx*CLI>
  1039.  -- Goto (macro-hangupcall,s,3)
  1040.  
  1041. pbx*CLI>
  1042.  -- Executing [s@macro-hangupcall:3] Hangup("SIP/6477232824-0000003e", "") in new stack
  1043.  
  1044. pbx*CLI>
  1045.  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/6477232824-0000003e' in macro 'hangupcall'
  1046.  
  1047. pbx*CLI>
  1048. Scheduling destruction of SIP dialog '3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7' in 6400 ms (Method: INVITE)
  1049.  
  1050. pbx*CLI>
  1051. set_destination: Parsing <sip:203@10.10.10.12:5060> for address/port to send to
  1052. set_destination: set destination to 10.10.10.12, port 5060
  1053.  
  1054. pbx*CLI>
  1055. Reliably Transmitting (NAT) to 10.10.10.12:5060:
  1056. BYE sip:203@10.10.10.12:5060 SIP/2.0
  1057.  
  1058. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK01ecde6f;rport
  1059.  
  1060. Max-Forwards: 70
  1061.  
  1062. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  1063.  
  1064. To: <sip:203@10.10.10.12:5060>;tag=e52b3a3cb5e9dd53i0
  1065.  
  1066. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  1067.  
  1068. CSeq: 103 BYE
  1069.  
  1070. User-Agent: FPBX-2.9.0(1.6.2.15)
  1071.  
  1072. X-Asterisk-HangupCause: Normal Clearing
  1073.  
  1074. X-Asterisk-HangupCauseCode: 16
  1075.  
  1076. Content-Length: 0
  1077.  
  1078.  
  1079.  
  1080.  
  1081. ---
  1082.  
  1083. pbx*CLI>
  1084.  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/6477232824-0000003e' in macro 'dial'
  1085.  
  1086. pbx*CLI>
  1087.  == Spawn extension (ext-group, 601, 12) exited non-zero on 'SIP/6477232824-0000003e'
  1088.  
  1089. pbx*CLI>
  1090. Scheduling destruction of SIP dialog '33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51' in 6400 ms (Method: ACK)
  1091.  
  1092. pbx*CLI>
  1093. set_destination: Parsing <sip:4168853558@216.58.0.51> for address/port to send to
  1094. set_destination: set destination to 216.58.0.51, port 5060
  1095.  
  1096. pbx*CLI>
  1097. Reliably Transmitting (no NAT) to 216.58.0.51:5060:
  1098. BYE sip:4168853558@216.58.0.51 SIP/2.0
  1099.  
  1100. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK199e2b57;rport
  1101.  
  1102. Max-Forwards: 70
  1103.  
  1104. From: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  1105.  
  1106. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  1107.  
  1108. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  1109.  
  1110. CSeq: 102 BYE
  1111.  
  1112. User-Agent: FPBX-2.9.0(1.6.2.15)
  1113.  
  1114. X-Asterisk-HangupCause: Normal Clearing
  1115.  
  1116. X-Asterisk-HangupCauseCode: 16
  1117.  
  1118. Content-Length: 0
  1119.  
  1120.  
  1121.  
  1122.  
  1123. ---
  1124.  
  1125. pbx*CLI>
  1126. 
  1127. <--- SIP read from UDP:10.10.10.12:5060 --->
  1128. SIP/2.0 200 OK
  1129.  
  1130. To: <sip:203@10.10.10.12:5060>;tag=e52b3a3cb5e9dd53i0
  1131.  
  1132. From: "Unknown" <sip:4168853558@10.10.10.7>;tag=as5693c08b
  1133.  
  1134. Call-ID: 3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7
  1135.  
  1136. CSeq: 103 BYE
  1137.  
  1138. Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK01ecde6f
  1139.  
  1140. Server: Linksys/SPA942-6.1.5(a)
  1141.  
  1142. Content-Length: 0
  1143.  
  1144.  
  1145.  
  1146.  
  1147. <------------->
  1148.  
  1149. pbx*CLI>
  1150. --- (8 headers 0 lines) ---
  1151.  
  1152. pbx*CLI>
  1153. Really destroying SIP dialog '3b4cd664420fa05e62bab75d28b07ca0@10.10.10.7' Method: INVITE
  1154.  
  1155. pbx*CLI>
  1156. Retransmitting #1 (no NAT) to 216.58.0.51:5060:
  1157. BYE sip:4168853558@216.58.0.51 SIP/2.0
  1158.  
  1159. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK199e2b57;rport
  1160.  
  1161. Max-Forwards: 70
  1162.  
  1163. From: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  1164.  
  1165. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  1166.  
  1167. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  1168.  
  1169. CSeq: 102 BYE
  1170.  
  1171. User-Agent: FPBX-2.9.0(1.6.2.15)
  1172.  
  1173. X-Asterisk-HangupCause: Normal Clearing
  1174.  
  1175. X-Asterisk-HangupCauseCode: 16
  1176.  
  1177. Content-Length: 0
  1178.  
  1179.  
  1180.  
  1181.  
  1182. ---
  1183.  
  1184. pbx*CLI>
  1185. 
  1186. <--- SIP read from UDP:216.58.0.51:5060 --->
  1187. SIP/2.0 200 OK
  1188.  
  1189. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK199e2b57;received=99.227.42.4;rport=5060
  1190.  
  1191. From: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  1192.  
  1193. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  1194.  
  1195. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  1196.  
  1197. CSeq: 102 BYE
  1198.  
  1199. User-Agent: CIA.com PBX
  1200.  
  1201. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1202.  
  1203. Supported: replaces
  1204.  
  1205. Contact: <sip:4168853558@216.58.0.51>
  1206.  
  1207. Content-Length: 0
  1208.  
  1209.  
  1210.  
  1211.  
  1212. <------------->
  1213.  
  1214. pbx*CLI>
  1215. --- (11 headers 0 lines) ---
  1216.  
  1217. pbx*CLI>
  1218. SIP Response message for INCOMING dialog BYE arrived
  1219.  
  1220. pbx*CLI>
  1221. Really destroying SIP dialog '33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51' Method: ACK
  1222.  
  1223. pbx*CLI>
  1224. 
  1225. <--- SIP read from UDP:216.58.0.51:5060 --->
  1226. SIP/2.0 200 OK
  1227.  
  1228. Via: SIP/2.0/UDP 99.227.42.4:5060;branch=z9hG4bK199e2b57;received=99.227.42.4;rport=5060
  1229.  
  1230. From: <sip:6477232824@99.227.42.4>;tag=as062a0d07
  1231.  
  1232. To: "Unknown" <sip:4168853558@216.58.0.51>;tag=as2b1ad912
  1233.  
  1234. Call-ID: 33fa7dc24040e1a8674c752c0a7176d7@216.58.0.51
  1235.  
  1236. CSeq: 102 BYE
  1237.  
  1238. User-Agent: CIA.com PBX
  1239.  
  1240. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1241.  
  1242. Supported: replaces
  1243.  
  1244. Contact: <sip:4168853558@216.58.0.51>
  1245.  
  1246. Content-Length: 0
  1247.  
  1248.  
  1249.  
  1250.  
  1251. <------------->
  1252.  
  1253. pbx*CLI>
  1254. --- (11 headers 0 lines) ---
  1255.  
  1256. pbx*CLI> exit
  1257.  
  1258. Executing last minute cleanups
  1259. ]0;root@pbx:~[root@pbx ~]# exit
  1260. logout
  1261. 
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