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  1. Asterisk 1.8.20.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  2. Created by Mark Spencer <markster@digium.com>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 1.8.20.0 currently running on pbxhmo (pid = 17641)
  9. pbxhmo*CLI>
  10. Verbosity is at least 3
  11.  
  12. pbxhmo*CLI>
  13. <--- SIP read from UDP:10.6.58.195:5060 --->
  14. INVITE sip:2156@10.11.3.9:5060 SIP/2.0
  15. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
  16. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  17. To: <sip:034750386625002156@10.11.3.9:5060>
  18. Call-ID: 67160462_45129454@10.6.58.195
  19. CSeq: 2711 INVITE
  20. Max-Forwards: 70
  21. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
  22. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
  23. Contact: <sip:6699156000@10.6.58.195:5060>
  24. Supported: timer,100rel
  25. Session-Expires: 1800
  26. Min-SE: 90
  27. Content-Length: 259
  28. Content-Disposition: session; handling=required
  29. Content-Type: application/sdp
  30.  
  31. v=0
  32. o=Sonus_UAC 17433 30551 IN IP4 10.6.58.195
  33. s=SIP Media Capabilities
  34. c=IN IP4 10.6.58.197
  35. t=0 0
  36. m=audio 8434 RTP/AVP 0 18 100
  37. a=rtpmap:0 PCMU/8000
  38. a=rtpmap:18 G729/8000
  39. a=rtpmap:100 telephone-event/8000
  40. a=fmtp:100 0-15
  41. a=sendrecv
  42. a=maxptime:20
  43. <------------->
  44.  
  45. pbxhmo*CLI>
  46. --- (16 headers 12 lines) ---
  47.  
  48. pbxhmo*CLI>
  49. Sending to 10.6.58.195:5060 (NAT)
  50. Using INVITE request as basis request - 67160462_45129454@10.6.58.195
  51.  
  52. pbxhmo*CLI>
  53. Found peer 'alestra' for '6699156000' from 10.6.58.195:5060
  54.  
  55. pbxhmo*CLI>
  56. == Using SIP RTP TOS bits 184
  57.  
  58. pbxhmo*CLI>
  59. == Using SIP RTP CoS mark 5
  60.  
  61. pbxhmo*CLI>
  62. Found RTP audio format 0
  63.  
  64. pbxhmo*CLI>
  65. Found RTP audio format 18
  66.  
  67. pbxhmo*CLI>
  68. Found RTP audio format 100
  69. Found audio description format PCMU for ID 0
  70.  
  71. pbxhmo*CLI>
  72. Found audio description format G729 for ID 18
  73.  
  74. pbxhmo*CLI>
  75. Found audio description format telephone-event for ID 100
  76.  
  77. pbxhmo*CLI>
  78. Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  79. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  80.  
  81. pbxhmo*CLI>
  82. Peer audio RTP is at port 10.6.58.197:8434
  83.  
  84. pbxhmo*CLI>
  85. Looking for 2156 in from-trunk-sip-alestra (domain 10.11.3.9)
  86.  
  87. pbxhmo*CLI>
  88. list_route: hop: <sip:6699156000@10.6.58.195:5060>
  89.  
  90. <--- Transmitting (NAT) to 10.6.58.195:5060 --->
  91. SIP/2.0 100 Trying
  92. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
  93. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  94. To: <sip:034750386625002156@10.11.3.9:5060>
  95. Call-ID: 67160462_45129454@10.6.58.195
  96. CSeq: 2711 INVITE
  97. Server: FPBX-2.8.1(1.8.20.0)
  98. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  99. Supported: replaces, timer
  100. Session-Expires: 1800;refresher=uas
  101. Contact: <sip:2156@10.11.3.9:5060>
  102. Content-Length: 0
  103.  
  104.  
  105. <------------>
  106. -- Executing [2156@from-trunk-sip-alestra:1] Set("SIP/alestra-0000008a", "GROUP()=OUT_2") in new stack
  107. -- Executing [2156@from-trunk-sip-alestra:2] Goto("SIP/alestra-0000008a", "from-trunk,2156,1") in new stack
  108. -- Goto (from-trunk,2156,1)
  109. -- Executing [2156@from-trunk:1] Set("SIP/alestra-0000008a", "__FROM_DID=2156") in new stack
  110. -- Executing [2156@from-trunk:2] Gosub("SIP/alestra-0000008a", "app-blacklist-check,s,1") in new stack
  111. -- Executing [s@app-blacklist-check:1] GotoIf("SIP/alestra-0000008a", "0?blacklisted") in new stack
  112. -- Executing [s@app-blacklist-check:2] Set("SIP/alestra-0000008a", "CALLED_BLACKLIST=1") in new stack
  113. -- Executing [s@app-blacklist-check:3] Return("SIP/alestra-0000008a", "") in new stack
  114. -- Executing [2156@from-trunk:3] ExecIf("SIP/alestra-0000008a", "1 ?Set(CALLERID(name)=6699156000)") in new stack
  115. -- Executing [2156@from-trunk:4] Set("SIP/alestra-0000008a", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  116. -- Executing [2156@from-trunk:5] Set("SIP/alestra-0000008a", "CALLERPRES()=allowed_not_screened") in new stack
  117. -- Executing [2156@from-trunk:6] Goto("SIP/alestra-0000008a", "from-did-direct,104,1") in new stack
  118. -- Goto (from-did-direct,104,1)
  119. -- Executing [104@from-did-direct:1] Macro("SIP/alestra-0000008a", "exten-vm,novm,104") in new stack
  120. -- Executing [s@macro-exten-vm:1] Macro("SIP/alestra-0000008a", "user-callerid,") in new stack
  121. -- Executing [s@macro-user-callerid:1] Set("SIP/alestra-0000008a", "AMPUSER=6699156000") in new stack
  122. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/alestra-0000008a", "0?report") in new stack
  123. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/alestra-0000008a", "1?Set(REALCALLERIDNUM=6699156000)") in new stack
  124. -- Executing [s@macro-user-callerid:4] Set("SIP/alestra-0000008a", "AMPUSER=") in new stack
  125. -- Executing [s@macro-user-callerid:5] Set("SIP/alestra-0000008a", "AMPUSERCIDNAME=") in new stack
  126. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/alestra-0000008a", "1?report") in new stack
  127. -- Goto (macro-user-callerid,s,10)
  128. -- Executing [s@macro-user-callerid:10] GotoIf("SIP/alestra-0000008a", "0?continue") in new stack
  129. -- Executing [s@macro-user-callerid:11] Set("SIP/alestra-0000008a", "__TTL=64") in new stack
  130. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/alestra-0000008a", "1?continue") in new stack
  131. -- Goto (macro-user-callerid,s,19)
  132. -- Executing [s@macro-user-callerid:19] Set("SIP/alestra-0000008a", "CALLERID(number)=6699156000") in new stack
  133. -- Executing [s@macro-user-callerid:20] Set("SIP/alestra-0000008a", "CALLERID(name)=6699156000") in new stack
  134. -- Executing [s@macro-user-callerid:21] NoOp("SIP/alestra-0000008a", "Using CallerID "6699156000" <6699156000>") in new stack
  135. -- Executing [s@macro-exten-vm:2] Set("SIP/alestra-0000008a", "RingGroupMethod=none") in new stack
  136. -- Executing [s@macro-exten-vm:3] Set("SIP/alestra-0000008a", "VMBOX=novm") in new stack
  137. -- Executing [s@macro-exten-vm:4] Set("SIP/alestra-0000008a", "__EXTTOCALL=104") in new stack
  138. -- Executing [s@macro-exten-vm:5] Set("SIP/alestra-0000008a", "CFUEXT=") in new stack
  139. -- Executing [s@macro-exten-vm:6] Set("SIP/alestra-0000008a", "CFBEXT=") in new stack
  140. -- Executing [s@macro-exten-vm:7] Set("SIP/alestra-0000008a", "RT=""") in new stack
  141. -- Executing [s@macro-exten-vm:8] Macro("SIP/alestra-0000008a", "record-enable,104,IN") in new stack
  142.  
  143. pbxhmo*CLI>
  144. -- Executing [s@macro-record-enable:1] GotoIf("SIP/alestra-0000008a", "1?check") in new stack
  145. -- Goto (macro-record-enable,s,4)
  146. -- Executing [s@macro-record-enable:4] ExecIf("SIP/alestra-0000008a", "0?MacroExit()") in new stack
  147.  
  148. pbxhmo*CLI>
  149. -- Executing [s@macro-record-enable:5] GotoIf("SIP/alestra-0000008a", "0?Group:OUT") in new stack
  150.  
  151. pbxhmo*CLI>
  152. -- Goto (macro-record-enable,s,15)
  153.  
  154. pbxhmo*CLI>
  155. -- Executing [s@macro-record-enable:15] GotoIf("SIP/alestra-0000008a", "1?IN") in new stack
  156.  
  157. pbxhmo*CLI>
  158. -- Goto (macro-record-enable,s,20)
  159.  
  160. pbxhmo*CLI>
  161. -- Executing [s@macro-record-enable:20] ExecIf("SIP/alestra-0000008a", "1?MacroExit()") in new stack
  162.  
  163. pbxhmo*CLI>
  164. -- Executing [s@macro-exten-vm:9] Macro("SIP/alestra-0000008a", "dial-one,"",tr,104") in new stack
  165.  
  166. pbxhmo*CLI>
  167. -- Executing [s@macro-dial-one:1] Set("SIP/alestra-0000008a", "DEXTEN=104") in new stack
  168.  
  169. pbxhmo*CLI>
  170. -- Executing [s@macro-dial-one:2] Set("SIP/alestra-0000008a", "DIALSTATUS_CW=") in new stack
  171.  
  172. pbxhmo*CLI>
  173. -- Executing [s@macro-dial-one:3] GosubIf("SIP/alestra-0000008a", "0?screen,1") in new stack
  174.  
  175. pbxhmo*CLI>
  176. -- Executing [s@macro-dial-one:4] GosubIf("SIP/alestra-0000008a", "0?cf,1") in new stack
  177.  
  178. pbxhmo*CLI>
  179. -- Executing [s@macro-dial-one:5] GotoIf("SIP/alestra-0000008a", "1?skip1") in new stack
  180.  
  181. pbxhmo*CLI>
  182. -- Goto (macro-dial-one,s,8)
  183.  
  184. pbxhmo*CLI>
  185. -- Executing [s@macro-dial-one:8] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack
  186.  
  187. pbxhmo*CLI>
  188. -- Executing [s@macro-dial-one:9] GotoIf("SIP/alestra-0000008a", "0?continue") in new stack
  189.  
  190. pbxhmo*CLI>
  191. -- Executing [s@macro-dial-one:10] Set("SIP/alestra-0000008a", "EXTHASCW=") in new stack
  192.  
  193. pbxhmo*CLI>
  194. -- Executing [s@macro-dial-one:11] GotoIf("SIP/alestra-0000008a", "1?next1:cwinusebusy") in new stack
  195.  
  196. pbxhmo*CLI>
  197. -- Goto (macro-dial-one,s,12)
  198.  
  199. pbxhmo*CLI>
  200. -- Executing [s@macro-dial-one:12] GotoIf("SIP/alestra-0000008a", "0?docfu:skip3") in new stack
  201.  
  202. pbxhmo*CLI>
  203. -- Goto (macro-dial-one,s,16)
  204.  
  205. pbxhmo*CLI>
  206. -- Executing [s@macro-dial-one:16] GotoIf("SIP/alestra-0000008a", "1?next2:continue") in new stack
  207.  
  208. pbxhmo*CLI>
  209. -- Goto (macro-dial-one,s,17)
  210.  
  211. pbxhmo*CLI>
  212. -- Executing [s@macro-dial-one:17] GotoIf("SIP/alestra-0000008a", "1?continue") in new stack
  213.  
  214. pbxhmo*CLI>
  215. -- Goto (macro-dial-one,s,25)
  216.  
  217. pbxhmo*CLI>
  218. -- Executing [s@macro-dial-one:25] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack
  219.  
  220. pbxhmo*CLI>
  221. -- Executing [s@macro-dial-one:26] GosubIf("SIP/alestra-0000008a", "1?dstring,1:dlocal,1") in new stack
  222.  
  223. pbxhmo*CLI>
  224. -- Executing [dstring@macro-dial-one:1] Set("SIP/alestra-0000008a", "DSTRING=") in new stack
  225.  
  226. pbxhmo*CLI>
  227. -- Executing [dstring@macro-dial-one:2] Set("SIP/alestra-0000008a", "DEVICES=104") in new stack
  228.  
  229. pbxhmo*CLI>
  230. -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/alestra-0000008a", "0?Return()") in new stack
  231.  
  232. pbxhmo*CLI>
  233. -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/alestra-0000008a", "0?Set(DEVICES=04)") in new stack
  234.  
  235. pbxhmo*CLI>
  236. -- Executing [dstring@macro-dial-one:5] Set("SIP/alestra-0000008a", "LOOPCNT=1") in new stack
  237.  
  238. pbxhmo*CLI>
  239. -- Executing [dstring@macro-dial-one:6] Set("SIP/alestra-0000008a", "ITER=1") in new stack
  240.  
  241. pbxhmo*CLI>
  242. -- Executing [dstring@macro-dial-one:7] Set("SIP/alestra-0000008a", "THISDIAL=SIP/104") in new stack
  243.  
  244. pbxhmo*CLI>
  245. -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/alestra-0000008a", "1?zap2dahdi,1") in new stack
  246.  
  247. pbxhmo*CLI>
  248. -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/alestra-0000008a", "0?Return()") in new stack
  249.  
  250. pbxhmo*CLI>
  251. -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/alestra-0000008a", "NEWDIAL=") in new stack
  252.  
  253. pbxhmo*CLI>
  254. -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/alestra-0000008a", "LOOPCNT2=1") in new stack
  255.  
  256. pbxhmo*CLI>
  257. -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/alestra-0000008a", "ITER2=1") in new stack
  258.  
  259. pbxhmo*CLI>
  260. -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/alestra-0000008a", "THISPART2=SIP/104") in new stack
  261.  
  262. pbxhmo*CLI>
  263. -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/alestra-0000008a", "0?Set(THISPART2=DAHDI/104)") in new stack
  264.  
  265. pbxhmo*CLI>
  266. -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/alestra-0000008a", "NEWDIAL=SIP/104&") in new stack
  267.  
  268. pbxhmo*CLI>
  269. -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/alestra-0000008a", "ITER2=2") in new stack
  270.  
  271. pbxhmo*CLI>
  272. -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/alestra-0000008a", "0?begin2") in new stack
  273.  
  274. pbxhmo*CLI>
  275. -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/alestra-0000008a", "THISDIAL=SIP/104") in new stack
  276.  
  277. pbxhmo*CLI>
  278. -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/alestra-0000008a", "") in new stack
  279.  
  280. pbxhmo*CLI>
  281. -- Executing [dstring@macro-dial-one:9] Set("SIP/alestra-0000008a", "DSTRING=SIP/104&") in new stack
  282.  
  283. pbxhmo*CLI>
  284. -- Executing [dstring@macro-dial-one:10] Set("SIP/alestra-0000008a", "ITER=2") in new stack
  285.  
  286. pbxhmo*CLI>
  287. -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/alestra-0000008a", "0?begin") in new stack
  288.  
  289. pbxhmo*CLI>
  290. -- Executing [dstring@macro-dial-one:12] Set("SIP/alestra-0000008a", "DSTRING=SIP/104") in new stack
  291.  
  292. pbxhmo*CLI>
  293. -- Executing [dstring@macro-dial-one:13] Return("SIP/alestra-0000008a", "") in new stack
  294.  
  295. pbxhmo*CLI>
  296. -- Executing [s@macro-dial-one:27] GotoIf("SIP/alestra-0000008a", "0?nodial") in new stack
  297.  
  298. pbxhmo*CLI>
  299. -- Executing [s@macro-dial-one:28] GotoIf("SIP/alestra-0000008a", "0?skiptrace") in new stack
  300.  
  301. pbxhmo*CLI>
  302. -- Executing [s@macro-dial-one:29] GosubIf("SIP/alestra-0000008a", "1?ctset,1:ctclear,1") in new stack
  303.  
  304. pbxhmo*CLI>
  305. -- Executing [ctset@macro-dial-one:1] Set("SIP/alestra-0000008a", "DB(CALLTRACE/104)=6699156000") in new stack
  306.  
  307. pbxhmo*CLI>
  308. -- Executing [ctset@macro-dial-one:2] Return("SIP/alestra-0000008a", "") in new stack
  309.  
  310. pbxhmo*CLI>
  311. -- Executing [s@macro-dial-one:30] Set("SIP/alestra-0000008a", "D_OPTIONS=tr") in new stack
  312.  
  313. pbxhmo*CLI>
  314. -- Executing [s@macro-dial-one:31] ExecIf("SIP/alestra-0000008a", "0?SIPAddHeader(Alert-Info: )") in new stack
  315.  
  316. pbxhmo*CLI>
  317. -- Executing [s@macro-dial-one:32] ExecIf("SIP/alestra-0000008a", "0?SIPAddHeader()") in new stack
  318.  
  319. pbxhmo*CLI>
  320. -- Executing [s@macro-dial-one:33] ExecIf("SIP/alestra-0000008a", "0?Set(CHANNEL(musicclass)=)") in new stack
  321.  
  322. pbxhmo*CLI>
  323. -- Executing [s@macro-dial-one:34] GosubIf("SIP/alestra-0000008a", "0?qwait,1") in new stack
  324.  
  325. pbxhmo*CLI>
  326. -- Executing [s@macro-dial-one:35] Set("SIP/alestra-0000008a", "__CWIGNORE=") in new stack
  327.  
  328. pbxhmo*CLI>
  329. -- Executing [s@macro-dial-one:36] Set("SIP/alestra-0000008a", "__KEEPCID=TRUE") in new stack
  330.  
  331. pbxhmo*CLI>
  332. -- Executing [s@macro-dial-one:37] Dial("SIP/alestra-0000008a", "SIP/104,"",tr") in new stack
  333.  
  334. pbxhmo*CLI>
  335. == Using SIP RTP TOS bits 184
  336.  
  337. pbxhmo*CLI>
  338. == Using SIP RTP CoS mark 5
  339.  
  340. pbxhmo*CLI>
  341. Audio is at 13306
  342.  
  343. pbxhmo*CLI>
  344. Adding codec 0x4 (ulaw) to SDP
  345.  
  346. pbxhmo*CLI>
  347. Adding codec 0x8 (alaw) to SDP
  348. Adding codec 0x2 (gsm) to SDP
  349.  
  350. pbxhmo*CLI>
  351. Adding non-codec 0x1 (telephone-event) to SDP
  352.  
  353. pbxhmo*CLI>
  354. Reliably Transmitting (NAT) to 10.11.1.153:5062:
  355. INVITE sip:104@10.11.1.153:5062 SIP/2.0
  356. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  357. Max-Forwards: 70
  358. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  359. To: <sip:104@10.11.1.153:5062>
  360. Contact: <sip:6699156000@10.11.1.9:5060>
  361. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  362. CSeq: 102 INVITE
  363. User-Agent: FPBX-2.8.1(1.8.20.0)
  364. Date: Tue, 31 Dec 2013 20:17:23 GMT
  365. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  366. Supported: replaces, timer
  367. Content-Type: application/sdp
  368. Content-Length: 276
  369.  
  370. v=0
  371. o=root 241784902 241784902 IN IP4 10.11.1.9
  372. s=Asterisk PBX 1.8.20.0
  373. c=IN IP4 10.11.1.9
  374. t=0 0
  375. m=audio 13306 RTP/AVP 0 8 3 101
  376. a=rtpmap:0 PCMU/8000
  377. a=rtpmap:8 PCMA/8000
  378. a=rtpmap:3 GSM/8000
  379. a=rtpmap:101 telephone-event/8000
  380. a=fmtp:101 0-16
  381. a=ptime:20
  382. a=sendrecv
  383.  
  384. ---
  385.  
  386. pbxhmo*CLI>
  387. -- Called SIP/104
  388.  
  389. pbxhmo*CLI>
  390. <--- Transmitting (NAT) to 10.6.58.195:5060 --->
  391. SIP/2.0 180 Ringing
  392. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
  393. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  394. To: <sip:034750386625002156@10.11.3.9:5060>;tag=as686231eb
  395. Call-ID: 67160462_45129454@10.6.58.195
  396. CSeq: 2711 INVITE
  397. Server: FPBX-2.8.1(1.8.20.0)
  398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  399. Supported: replaces, timer
  400. Session-Expires: 1800;refresher=uas
  401. Contact: <sip:2156@10.11.3.9:5060>
  402. Content-Length: 0
  403.  
  404.  
  405. <------------>
  406.  
  407. pbxhmo*CLI>
  408. <--- SIP read from UDP:10.11.1.153:5062 --->
  409. SIP/2.0 100 Trying
  410. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  411. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  412. To: <sip:104@10.11.1.153:5062>
  413. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  414. CSeq: 102 INVITE
  415. User-Agent: Yealink SIP-T20P 9.71.0.168
  416. Content-Length: 0
  417.  
  418. <------------->
  419.  
  420. pbxhmo*CLI>
  421. --- (8 headers 0 lines) ---
  422.  
  423. pbxhmo*CLI>
  424. <--- SIP read from UDP:10.11.1.153:5062 --->
  425. SIP/2.0 180 Ringing
  426. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  427. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  428. To: <sip:104@10.11.1.153:5062>;tag=973490739
  429. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  430. CSeq: 102 INVITE
  431. Contact: <sip:104@10.11.1.153:5062>
  432. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  433. User-Agent: Yealink SIP-T20P 9.71.0.168
  434. Allow-Events: talk,hold,conference,refer,check-sync
  435. Content-Length: 0
  436.  
  437. <------------->
  438.  
  439. pbxhmo*CLI>
  440. --- (11 headers 0 lines) ---
  441.  
  442. pbxhmo*CLI>
  443. list_route: hop: <sip:104@10.11.1.153:5062>
  444.  
  445. pbxhmo*CLI>
  446. -- SIP/104-0000008b is ringing
  447.  
  448. pbxhmo*CLI>
  449. <--- Transmitting (NAT) to 10.6.58.195:5060 --->
  450. SIP/2.0 180 Ringing
  451. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
  452. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  453. To: <sip:034750386625002156@10.11.3.9:5060>;tag=as686231eb
  454. Call-ID: 67160462_45129454@10.6.58.195
  455. CSeq: 2711 INVITE
  456. Server: FPBX-2.8.1(1.8.20.0)
  457. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  458. Supported: replaces, timer
  459. Session-Expires: 1800;refresher=uas
  460. Contact: <sip:2156@10.11.3.9:5060>
  461. Content-Length: 0
  462.  
  463.  
  464. <------------>
  465.  
  466. pbxhmo*CLI>
  467. Reliably Transmitting (NAT) to 10.11.1.156:5062:
  468. OPTIONS sip:106@10.11.1.156:5062 SIP/2.0
  469. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK4de87984;rport
  470. Max-Forwards: 70
  471. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as68bcda31
  472. To: <sip:106@10.11.1.156:5062>
  473. Contact: <sip:Unknown@10.11.1.9:5060>
  474. Call-ID: 512991546113540413aa027a77f9688d@10.11.1.9:5060
  475. CSeq: 102 OPTIONS
  476. User-Agent: FPBX-2.8.1(1.8.20.0)
  477. Date: Tue, 31 Dec 2013 20:17:26 GMT
  478. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  479. Supported: replaces, timer
  480. Content-Length: 0
  481.  
  482.  
  483. ---
  484.  
  485. pbxhmo*CLI>
  486. <--- SIP read from UDP:10.11.1.156:5062 --->
  487.  
  488.  
  489. <------------->
  490.  
  491. pbxhmo*CLI>
  492. <--- SIP read from UDP:10.11.1.156:5062 --->
  493. SIP/2.0 200 OK
  494. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK4de87984;rport
  495. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as68bcda31
  496. To: <sip:106@10.11.1.156:5062>;tag=1463935576
  497. Call-ID: 512991546113540413aa027a77f9688d@10.11.1.9:5060
  498. CSeq: 102 OPTIONS
  499. User-Agent: Yealink SIP-T20P 9.71.0.168
  500. Content-Length: 0
  501.  
  502. <------------->
  503.  
  504. pbxhmo*CLI>
  505. --- (8 headers 0 lines) ---
  506.  
  507. pbxhmo*CLI>
  508. Really destroying SIP dialog '512991546113540413aa027a77f9688d@10.11.1.9:5060' Method: OPTIONS
  509.  
  510. pbxhmo*CLI>
  511. Reliably Transmitting (NAT) to 10.11.1.159:5062:
  512. OPTIONS sip:109@10.11.1.159:5062 SIP/2.0
  513. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK552f4af7;rport
  514. Max-Forwards: 70
  515. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as5b4d05c7
  516. To: <sip:109@10.11.1.159:5062>
  517. Contact: <sip:Unknown@10.11.1.9:5060>
  518. Call-ID: 1d5295c40fff743645a5ca9c3a86381e@10.11.1.9:5060
  519. CSeq: 102 OPTIONS
  520. User-Agent: FPBX-2.8.1(1.8.20.0)
  521. Date: Tue, 31 Dec 2013 20:17:29 GMT
  522. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  523. Supported: replaces, timer
  524. Content-Length: 0
  525.  
  526.  
  527. ---
  528.  
  529. pbxhmo*CLI>
  530. <--- SIP read from UDP:10.11.1.159:5062 --->
  531.  
  532.  
  533. <------------->
  534.  
  535. pbxhmo*CLI>
  536. <--- SIP read from UDP:10.11.1.159:5062 --->
  537. SIP/2.0 200 OK
  538. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK552f4af7;rport
  539. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as5b4d05c7
  540. To: <sip:109@10.11.1.159:5062>;tag=403368631
  541. Call-ID: 1d5295c40fff743645a5ca9c3a86381e@10.11.1.9:5060
  542. CSeq: 102 OPTIONS
  543. User-Agent: Yealink SIP-T20P 9.71.0.168
  544. Content-Length: 0
  545.  
  546. <------------->
  547.  
  548. pbxhmo*CLI>
  549. --- (8 headers 0 lines) ---
  550. Really destroying SIP dialog '1d5295c40fff743645a5ca9c3a86381e@10.11.1.9:5060' Method: OPTIONS
  551.  
  552. pbxhmo*CLI>
  553. Reliably Transmitting (NAT) to 10.11.1.164:5062:
  554. OPTIONS sip:114@10.11.1.164:5062 SIP/2.0
  555. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK180b13e9;rport
  556. Max-Forwards: 70
  557. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as7e7435fe
  558. To: <sip:114@10.11.1.164:5062>
  559. Contact: <sip:Unknown@10.11.1.9:5060>
  560. Call-ID: 09b62ecf752ae07b1ead007c400db1b2@10.11.1.9:5060
  561. CSeq: 102 OPTIONS
  562. User-Agent: FPBX-2.8.1(1.8.20.0)
  563. Date: Tue, 31 Dec 2013 20:17:29 GMT
  564. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  565. Supported: replaces, timer
  566. Content-Length: 0
  567.  
  568.  
  569. ---
  570.  
  571. pbxhmo*CLI>
  572. <--- SIP read from UDP:10.11.1.164:5062 --->
  573.  
  574.  
  575. <------------->
  576.  
  577. pbxhmo*CLI>
  578. <--- SIP read from UDP:10.11.1.164:5062 --->
  579. SIP/2.0 200 OK
  580. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK180b13e9;rport
  581. From: "Unknown" <sip:Unknown@10.11.1.9>;tag=as7e7435fe
  582. To: <sip:114@10.11.1.164:5062>;tag=1752043124
  583. Call-ID: 09b62ecf752ae07b1ead007c400db1b2@10.11.1.9:5060
  584. CSeq: 102 OPTIONS
  585. User-Agent: Yealink SIP-T20P 9.71.0.168
  586. Content-Length: 0
  587.  
  588. <------------->
  589.  
  590. pbxhmo*CLI>
  591. --- (8 headers 0 lines) ---
  592.  
  593. pbxhmo*CLI>
  594. Really destroying SIP dialog '09b62ecf752ae07b1ead007c400db1b2@10.11.1.9:5060' Method: OPTIONS
  595.  
  596. pbxhmo*CLI>
  597. <--- SIP read from UDP:10.11.1.154:5062 --->
  598.  
  599.  
  600. <------------->
  601.  
  602. pbxhmo*CLI>
  603. <--- SIP read from UDP:10.6.58.195:5060 --->
  604. CANCEL sip:2156@10.11.3.9:5060 SIP/2.0
  605. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
  606. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  607. To: <sip:034750386625002156@10.11.3.9:5060>
  608. Call-ID: 67160462_45129454@10.6.58.195
  609. CSeq: 2711 CANCEL
  610. Max-Forwards: 70
  611. Content-Length: 0
  612.  
  613. <------------->
  614. --- (8 headers 0 lines) ---
  615. Sending to 10.6.58.195:5060 (NAT)
  616.  
  617. <--- Reliably Transmitting (NAT) to 10.6.58.195:5060 --->
  618. SIP/2.0 487 Request Terminated
  619. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
  620. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  621. To: <sip:034750386625002156@10.11.3.9:5060>;tag=as686231eb
  622. Call-ID: 67160462_45129454@10.6.58.195
  623. CSeq: 2711 INVITE
  624. Server: FPBX-2.8.1(1.8.20.0)
  625. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  626. Supported: replaces, timer
  627. Content-Length: 0
  628.  
  629.  
  630. <------------>
  631.  
  632. pbxhmo*CLI>
  633. <--- Transmitting (NAT) to 10.6.58.195:5060 --->
  634. SIP/2.0 200 OK
  635. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3;received=10.6.58.195;rport=5060
  636. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  637. To: <sip:034750386625002156@10.11.3.9:5060>;tag=as686231eb
  638. Call-ID: 67160462_45129454@10.6.58.195
  639. CSeq: 2711 CANCEL
  640. Server: FPBX-2.8.1(1.8.20.0)
  641. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  642. Supported: replaces, timer
  643. Content-Length: 0
  644.  
  645.  
  646. <------------>
  647. Scheduling destruction of SIP dialog '4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060' in 6400 ms (Method: INVITE)
  648. Reliably Transmitting (NAT) to 10.11.1.153:5062:
  649. CANCEL sip:104@10.11.1.153:5062 SIP/2.0
  650. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  651. Max-Forwards: 70
  652. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  653. To: <sip:104@10.11.1.153:5062>
  654. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  655. CSeq: 102 CANCEL
  656. User-Agent: FPBX-2.8.1(1.8.20.0)
  657. Content-Length: 0
  658.  
  659.  
  660. ---
  661. Scheduling destruction of SIP dialog '4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060' in 6400 ms (Method: INVITE)
  662.  
  663. pbxhmo*CLI>
  664. == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/alestra-0000008a' in macro 'dial-one'
  665. == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/alestra-0000008a' in macro 'exten-vm'
  666. == Spawn extension (from-did-direct, 104, 1) exited non-zero on 'SIP/alestra-0000008a'
  667. -- Executing [h@from-did-direct:1] Macro("SIP/alestra-0000008a", "hangupcall,") in new stack
  668. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/alestra-0000008a", "1?endmixmoncheck") in new stack
  669. -- Goto (macro-hangupcall,s,9)
  670. -- Executing [s@macro-hangupcall:9] NoOp("SIP/alestra-0000008a", "End of MIXMON check") in new stack
  671. -- Executing [s@macro-hangupcall:10] GotoIf("SIP/alestra-0000008a", "1?nomeetmemon") in new stack
  672.  
  673. pbxhmo*CLI>
  674. -- Goto (macro-hangupcall,s,28)
  675.  
  676. pbxhmo*CLI>
  677. -- Executing [s@macro-hangupcall:28] NoOp("SIP/alestra-0000008a", "End of MEETME check") in new stack
  678.  
  679. pbxhmo*CLI>
  680. -- Executing [s@macro-hangupcall:29] GotoIf("SIP/alestra-0000008a", "1?noautomon") in new stack
  681.  
  682. pbxhmo*CLI>
  683. -- Goto (macro-hangupcall,s,34)
  684.  
  685. pbxhmo*CLI>
  686. -- Executing [s@macro-hangupcall:34] NoOp("SIP/alestra-0000008a", "TOUCH_MONITOR_OUTPUT=") in new stack
  687.  
  688. pbxhmo*CLI>
  689. -- Executing [s@macro-hangupcall:35] GotoIf("SIP/alestra-0000008a", "1?noautomon2") in new stack
  690.  
  691. pbxhmo*CLI>
  692. -- Goto (macro-hangupcall,s,41)
  693.  
  694. pbxhmo*CLI>
  695. -- Executing [s@macro-hangupcall:41] NoOp("SIP/alestra-0000008a", "MONITOR_FILENAME=") in new stack
  696.  
  697. pbxhmo*CLI>
  698. -- Executing [s@macro-hangupcall:42] GotoIf("SIP/alestra-0000008a", "1?skiprg") in new stack
  699.  
  700. pbxhmo*CLI>
  701. -- Goto (macro-hangupcall,s,45)
  702.  
  703. pbxhmo*CLI>
  704. -- Executing [s@macro-hangupcall:45] GotoIf("SIP/alestra-0000008a", "1?skipblkvm") in new stack
  705.  
  706. pbxhmo*CLI>
  707. -- Goto (macro-hangupcall,s,48)
  708.  
  709. pbxhmo*CLI>
  710. -- Executing [s@macro-hangupcall:48] GotoIf("SIP/alestra-0000008a", "1?theend") in new stack
  711.  
  712. pbxhmo*CLI>
  713. -- Goto (macro-hangupcall,s,50)
  714.  
  715. pbxhmo*CLI>
  716. -- Executing [s@macro-hangupcall:50] AGI("SIP/alestra-0000008a", "hangup.agi") in new stack
  717.  
  718. pbxhmo*CLI>
  719. -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
  720.  
  721. pbxhmo*CLI>
  722. <--- SIP read from UDP:10.11.1.153:5062 --->
  723. SIP/2.0 200 OK
  724. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  725. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  726. To: <sip:104@10.11.1.153:5062>;tag=973490739
  727. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  728. CSeq: 102 CANCEL
  729. User-Agent: Yealink SIP-T20P 9.71.0.168
  730. Content-Length: 0
  731.  
  732. <------------->
  733. --- (8 headers 0 lines) ---
  734.  
  735. pbxhmo*CLI>
  736. <--- SIP read from UDP:10.11.1.153:5062 --->
  737. SIP/2.0 487 Request Terminated
  738. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  739. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  740. To: <sip:104@10.11.1.153:5062>;tag=973490739
  741. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  742. CSeq: 102 INVITE
  743. User-Agent: Yealink SIP-T20P 9.71.0.168
  744. Content-Length: 0
  745.  
  746. <------------->
  747. --- (8 headers 0 lines) ---
  748. Transmitting (NAT) to 10.11.1.153:5062:
  749. ACK sip:104@10.11.1.153:5062 SIP/2.0
  750. Via: SIP/2.0/UDP 10.11.1.9:5060;branch=z9hG4bK30946f68;rport
  751. Max-Forwards: 70
  752. From: "6699156000" <sip:6699156000@10.11.1.9>;tag=as35a7e686
  753. To: <sip:104@10.11.1.153:5062>;tag=973490739
  754. Contact: <sip:6699156000@10.11.1.9:5060>
  755. Call-ID: 4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060
  756. CSeq: 102 ACK
  757. User-Agent: FPBX-2.8.1(1.8.20.0)
  758. Content-Length: 0
  759.  
  760.  
  761. ---
  762. Scheduling destruction of SIP dialog '4e1fe0d1037decfd08517a716fc014bd@10.11.1.9:5060' in 6400 ms (Method: INVITE)
  763.  
  764. pbxhmo*CLI>
  765. <--- SIP read from UDP:10.6.58.195:5060 --->
  766. ACK sip:2156@10.11.3.9:5060 SIP/2.0
  767. Via: SIP/2.0/UDP 10.6.58.195:5060;branch=z9hG4bK00B7805377874e178f3
  768. From: <sip:6699156000@10.6.58.195:5060;otg=HERTMXMOR_1002>;tag=gK0077c850
  769. To: <sip:034750386625002156@10.11.3.9:5060>;tag=as686231eb
  770. Call-ID: 67160462_45129454@10.6.58.195
  771. CSeq: 2711 ACK
  772. Max-Forwards: 70
  773. Content-Length: 0
  774.  
  775. <------------->
  776.  
  777. pbxhmo*CLI>
  778. --- (8 headers 0 lines) ---
  779.  
  780. pbxhmo*CLI>
  781. -- <SIP/alestra-0000008a>AGI Script hangup.agi completed, returning 0
  782.  
  783. pbxhmo*CLI>
  784. -- Executing [s@macro-hangupcall:51] Hangup("SIP/alestra-0000008a", "") in new stack
  785.  
  786. pbxhmo*CLI>
  787. == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/alestra-0000008a' in macro 'hangupcall'
  788.  
  789. pbxhmo*CLI>
  790. == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/alestra-0000008a'
  791.  
  792. pbxhmo*CLI>
  793. Really destroying SIP dialog '67160462_45129454@10.6.58.195' Method: ACK
  794.  
  795. pbxhmo*CLI>
  796. <--- SIP read from UDP:10.11.1.158:5062 --->
  797.  
  798.  
  799. <------------->
  800.  
  801. pbxhmo*CLI>
  802. Disconnected from Asterisk server
  803. Executing last minute cleanups
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