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  1. Asterisk 1.6.0.10-FONCORE-r40, Copyright (C) 1999 - 2008 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 1.6.0.10-FONCORE-r40 currently running on trixbox1 (pid = 2720)
  9. trixbox1*CLI>
  10. Verbosity is at least 4
  11. Core debug is at least 4
  12.  
  13. trixbox1*CLI>
  14. <--- SIP read from UDP://62.177.135.41:38383 --->
  15.  
  16. <------------->
  17.  
  18. trixbox1*CLI>
  19. <--- SIP read from UDP://62.177.135.41:38383 --->
  20.  
  21. <------------->
  22.  
  23. trixbox1*CLI>
  24. <--- SIP read from UDP://62.177.135.41:38383 --->
  25.  
  26. <------------->
  27.  
  28. trixbox1*CLI>
  29. == Manager 'admin' logged on from 127.0.0.1
  30.  
  31. trixbox1*CLI>
  32. <--- SIP read from UDP://62.177.135.41:38383 --->
  33. INVITE sip:s@195.241.40.168 SIP/2.0
  34. Record-Route: <sip:62.177.135.41:38383;ftag=as443a600e;lr=on>
  35. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK8bcd.d05ed0c1.0
  36. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK147bacdb
  37. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  38. To: <sip:31229707736@62.177.135.41:38383>
  39. Contact: <sip:private@62.177.135.40>
  40. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  41. CSeq: 102 INVITE
  42. User-Agent: bbned PBX
  43. Max-Forwards: 69
  44. Date: Tue, 22 Jun 2010 14:35:56 GMT
  45. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  46. Content-Type: application/sdp
  47. Content-Length: 263
  48.  
  49. v=0
  50. o=root 3592 3592 IN IP4 62.177.135.40
  51. s=session
  52. c=IN IP4 62.177.135.42
  53. t=0 0
  54. m=audio 46570 RTP/AVP 8 18 101
  55. a=rtpmap:8 PCMA/8000
  56. a=rtpmap:18 G729/8000
  57. a=fmtp:18 annexb=no
  58. a=rtpmap:101 telephone-event/8000
  59. a=fmtp:101 0-16
  60. a=silenceSupp:off - - - -
  61.  
  62. <------------->
  63. --- (15 headers 12 lines) ---
  64. == Using SIP RTP TOS bits 184
  65. == Using SIP RTP CoS mark 5
  66. == Using SIP VRTP TOS bits 136
  67. == Using SIP VRTP CoS mark 6
  68. Sending to 62.177.135.41 : 38383 (NAT)
  69. Using INVITE request as basis request - 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  70. No user 'private' in SIP users list
  71. Found peer 'Company-3' for 'private' from 62.177.135.41:38383
  72. Found RTP audio format 8
  73. Found RTP audio format 18
  74. Found RTP audio format 101
  75. Peer audio RTP is at port 62.177.135.42:46570
  76. Found audio description format PCMA for ID 8
  77. Found audio description format G729 for ID 18
  78. Got unsupported a:fmtp in SDP offer
  79. Found audio description format telephone-event for ID 101
  80. Got unsupported a:fmtp in SDP offer
  81. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  82. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  83. Peer audio RTP is at port 62.177.135.42:46570
  84. Looking for s in custom-fixdid (domain 195.241.40.168)
  85. list_route: hop: <sip:62.177.135.41:38383;ftag=as443a600e;lr=on>
  86.  
  87. <--- Transmitting (no NAT) to 62.177.135.41:38383 --->
  88. SIP/2.0 100 Trying
  89. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK8bcd.d05ed0c1.0;received=62.177.135.41
  90. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK147bacdb
  91. Record-Route: <sip:62.177.135.41:38383;ftag=as443a600e;lr=on>
  92. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  93. To: <sip:31229707736@62.177.135.41:38383>
  94. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  95. CSeq: 102 INVITE
  96. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  97. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  98. Supported: replaces, timer
  99. Contact: <sip:s@195.241.40.168>
  100. Content-Length: 0
  101.  
  102.  
  103. <------------>
  104. -- Executing [s@custom-fixdid:1] NoOp("SIP/31229707735-0a5a6ca0", "Fixing DID using information from SIP TO header") in new stack
  105. -- Executing [s@custom-fixdid:2] Set("SIP/31229707735-0a5a6ca0", "pseudodid=<sip:31229707736@62.177.135.41:38383>") in new stack
  106. -- Executing [s@custom-fixdid:3] Set("SIP/31229707735-0a5a6ca0", "pseudodid=<sip:31229707736") in new stack
  107. -- Executing [s@custom-fixdid:4] Set("SIP/31229707735-0a5a6ca0", "pseudodid=31229707736") in new stack
  108. -- Executing [s@custom-fixdid:5] Goto("SIP/31229707735-0a5a6ca0", "from-trunk,31229707736,1") in new stack
  109. -- Goto (from-trunk,31229707736,1)
  110. -- Executing [31229707736@from-trunk:1] Set("SIP/31229707735-0a5a6ca0", "__FROM_DID=31229707736") in new stack
  111. -- Executing [31229707736@from-trunk:2] Gosub("SIP/31229707735-0a5a6ca0", "app-blacklist-check,s,1") in new stack
  112. -- Executing [s@app-blacklist-check:1] GotoIf("SIP/31229707735-0a5a6ca0", "0?blacklisted") in new stack
  113. -- Executing [s@app-blacklist-check:2] Return("SIP/31229707735-0a5a6ca0", "") in new stack
  114. -- Executing [31229707736@from-trunk:3] ExecIf("SIP/31229707735-0a5a6ca0", "0 ?Set(CALLERID(name)=private)") in new stack
  115. -- Executing [31229707736@from-trunk:4] Set("SIP/31229707735-0a5a6ca0", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  116. -- Executing [31229707736@from-trunk:5] Set("SIP/31229707735-0a5a6ca0", "CALLERPRES()=allowed_not_screened") in new stack
  117. -- Executing [31229707736@from-trunk:6] Set("SIP/31229707735-0a5a6ca0", "_RGPREFIX=MDL") in new stack
  118. -- Executing [31229707736@from-trunk:7] Set("SIP/31229707735-0a5a6ca0", "CALLERID(name)=MDLanonymous") in new stack
  119. -- Executing [31229707736@from-trunk:8] Goto("SIP/31229707735-0a5a6ca0", "from-did-direct,200,1") in new stack
  120. -- Goto (from-did-direct,200,1)
  121. -- Executing [200@from-did-direct:1] GotoIf("SIP/31229707735-0a5a6ca0", "1?ext-local,200,1") in new stack
  122. -- Goto (ext-local,200,1)
  123. -- Executing [200@ext-local:1] Macro("SIP/31229707735-0a5a6ca0", "exten-vm,200,200") in new stack
  124. -- Executing [s@macro-exten-vm:1] Macro("SIP/31229707735-0a5a6ca0", "user-callerid") in new stack
  125. -- Executing [s@macro-user-callerid:1] Set("SIP/31229707735-0a5a6ca0", "AMPUSER=private") in new stack
  126. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/31229707735-0a5a6ca0", "0?report") in new stack
  127. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/31229707735-0a5a6ca0", "1?Set(REALCALLERIDNUM=private)") in new stack
  128.  
  129. trixbox1*CLI>
  130. -- Executing [s@macro-user-callerid:4] Set("SIP/31229707735-0a5a6ca0", "AMPUSER=") in new stack
  131. -- Executing [s@macro-user-callerid:5] Set("SIP/31229707735-0a5a6ca0", "AMPUSERCIDNAME=") in new stack
  132. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/31229707735-0a5a6ca0", "1?report") in new stack
  133. -- Goto (macro-user-callerid,s,11)
  134. -- Executing [s@macro-user-callerid:11] GotoIf("SIP/31229707735-0a5a6ca0", "0?continue") in new stack
  135. -- Executing [s@macro-user-callerid:12] Set("SIP/31229707735-0a5a6ca0", "__TTL=64") in new stack
  136. -- Executing [s@macro-user-callerid:13] GotoIf("SIP/31229707735-0a5a6ca0", "1?continue") in new stack
  137. -- Goto (macro-user-callerid,s,20)
  138. -- Executing [s@macro-user-callerid:20] NoOp("SIP/31229707735-0a5a6ca0", "Using CallerID "MDLanonymous" <private>") in new stack
  139. -- Executing [s@macro-exten-vm:2] Set("SIP/31229707735-0a5a6ca0", "RingGroupMethod=none") in new stack
  140. -- Executing [s@macro-exten-vm:3] Set("SIP/31229707735-0a5a6ca0", "VMBOX=200") in new stack
  141. -- Executing [s@macro-exten-vm:4] Set("SIP/31229707735-0a5a6ca0", "EXTTOCALL=200") in new stack
  142. -- Executing [s@macro-exten-vm:5] Set("SIP/31229707735-0a5a6ca0", "CFUEXT=") in new stack
  143. -- Executing [s@macro-exten-vm:6] Set("SIP/31229707735-0a5a6ca0", "CFBEXT=") in new stack
  144. -- Executing [s@macro-exten-vm:7] Set("SIP/31229707735-0a5a6ca0", "RT=15") in new stack
  145. -- Executing [s@macro-exten-vm:8] Macro("SIP/31229707735-0a5a6ca0", "record-enable,200,IN") in new stack
  146. -- Executing [s@macro-record-enable:1] GotoIf("SIP/31229707735-0a5a6ca0", "1?check") in new stack
  147. -- Goto (macro-record-enable,s,4)
  148. -- Executing [s@macro-record-enable:4] AGI("SIP/31229707735-0a5a6ca0", "recordingcheck,20100622-163556,1277217356.11") in new stack
  149. -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  150.  
  151. trixbox1*CLI>
  152. recordingcheck,20100622-163556,1277217356.11: Inbound recording not enabled
  153.  
  154. trixbox1*CLI>
  155. -- <SIP/31229707735-0a5a6ca0>AGI Script recordingcheck completed, returning 0
  156.  
  157. trixbox1*CLI>
  158. -- Executing [s@macro-record-enable:5] MacroExit("SIP/31229707735-0a5a6ca0", "") in new stack
  159. -- Executing [s@macro-exten-vm:9] Macro("SIP/31229707735-0a5a6ca0", "dial,15,tr,200") in new stack
  160.  
  161. trixbox1*CLI>
  162. -- Executing [s@macro-dial:1] GotoIf("SIP/31229707735-0a5a6ca0", "1?dial") in new stack
  163. -- Goto (macro-dial,s,3)
  164. -- Executing [s@macro-dial:3] AGI("SIP/31229707735-0a5a6ca0", "dialparties.agi") in new stack
  165.  
  166. trixbox1*CLI>
  167. -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  168.  
  169. trixbox1*CLI>
  170. dialparties.agi: Starting New Dialparties.agi
  171.  
  172. trixbox1*CLI>
  173. == Manager 'admin' logged on from 127.0.0.1
  174.  
  175. trixbox1*CLI>
  176. dialparties.agi: Caller ID name is 'MDLanonymous' number is 'private'
  177.  
  178. trixbox1*CLI>
  179. dialparties.agi: Methodology of ring is 'none'
  180.  
  181. trixbox1*CLI>
  182. -- dialparties.agi: Added extension 200 to extension map
  183.  
  184. trixbox1*CLI>
  185. > dialparties.agi: Extension 200 has call screening off
  186.  
  187. trixbox1*CLI>
  188. -- dialparties.agi: Extension 200 cf is disabled
  189.  
  190. trixbox1*CLI>
  191. -- dialparties.agi: Extension 200 do not disturb is disabled
  192.  
  193. trixbox1*CLI>
  194. > dialparties.agi: extnum 200 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  195.  
  196. trixbox1*CLI>
  197. == Manager 'admin' logged off from 127.0.0.1
  198.  
  199. trixbox1*CLI>
  200. > dialparties.agi: ExtensionState: 0
  201.  
  202. trixbox1*CLI>
  203. -- dialparties.agi: DbDel CALLTRACE/200 - Caller ID is not defined
  204.  
  205. trixbox1*CLI>
  206. -- dialparties.agi: Filtered ARG3: 200
  207.  
  208. trixbox1*CLI>
  209. == Manager 'admin' logged off from 127.0.0.1
  210.  
  211. trixbox1*CLI>
  212. -- <SIP/31229707735-0a5a6ca0>AGI Script dialparties.agi completed, returning 0
  213.  
  214. trixbox1*CLI>
  215. -- Executing [s@macro-dial:7] Dial("SIP/31229707735-0a5a6ca0", "SIP/200,15,tr") in new stack
  216. == Using SIP RTP TOS bits 184
  217. == Using SIP RTP CoS mark 5
  218. == Using SIP VRTP TOS bits 136
  219. == Using SIP VRTP CoS mark 6
  220. Audio is at 195.241.40.168 port 11000
  221. Video is at 195.241.40.168 port 12232
  222. Adding codec 0x8 (alaw) to SDP
  223. Adding codec 0x4 (ulaw) to SDP
  224. Adding video codec 0x80000 (h263) to SDP
  225. Adding video codec 0x200000 (h264) to SDP
  226. Adding non-codec 0x1 (telephone-event) to SDP
  227. Reliably Transmitting (NAT) to 85.223.99.38:41532:
  228. INVITE sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4 SIP/2.0
  229. Via: SIP/2.0/UDP 195.241.40.168:5060;branch=z9hG4bK51bd45ae;rport
  230. Max-Forwards: 70
  231. From: "MDLanonymous" <sip:private@195.241.40.168>;tag=as3a842f28
  232. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  233. Contact: <sip:private@195.241.40.168>
  234. Call-ID: 1850333112755b7129c2ce8f70a2a644@195.241.40.168
  235. CSeq: 102 INVITE
  236. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  237. Date: Tue, 22 Jun 2010 14:35:57 GMT
  238. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  239. Supported: replaces, timer
  240. Content-Type: application/sdp
  241. Content-Length: 403
  242.  
  243. v=0
  244. o=root 1179129330 1179129330 IN IP4 195.241.40.168
  245. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  246. c=IN IP4 195.241.40.168
  247. b=CT:384
  248. t=0 0
  249. m=audio 11000 RTP/AVP 8 0 101
  250. a=rtpmap:8 PCMA/8000
  251. a=rtpmap:0 PCMU/8000
  252. a=rtpmap:101 telephone-event/8000
  253. a=fmtp:101 0-16
  254. a=silenceSupp:off - - - -
  255. a=ptime:20
  256. a=sendrecv
  257. m=video 12232 RTP/AVP 34 99
  258. a=rtpmap:34 H263/90000
  259. a=rtpmap:99 H264/90000
  260. a=sendrecv
  261.  
  262. ---
  263. -- Called 200
  264.  
  265. <--- Transmitting (no NAT) to 62.177.135.41:38383 --->
  266. SIP/2.0 180 Ringing
  267. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK8bcd.d05ed0c1.0;received=62.177.135.41
  268. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK147bacdb
  269. Record-Route: <sip:62.177.135.41:38383;ftag=as443a600e;lr=on>
  270. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  271. To: <sip:31229707736@62.177.135.41:38383>;tag=as66f139a0
  272. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  273. CSeq: 102 INVITE
  274. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  275. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  276. Supported: replaces, timer
  277. Contact: <sip:s@195.241.40.168>
  278. Content-Length: 0
  279.  
  280.  
  281. <------------>
  282.  
  283. trixbox1*CLI>
  284. <--- SIP read from UDP://85.223.99.38:41532 --->
  285. SIP/2.0 180 Ringing
  286. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  287. Contact: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  288. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  289. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  290. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  291. CSeq: 102 INVITE
  292. User-Agent: X-Lite release 1002tx stamp 29712
  293. Content-Length: 0
  294.  
  295.  
  296. <------------->
  297. --- (9 headers 0 lines) ---
  298.  
  299. trixbox1*CLI>
  300. Reliably Transmitting (NAT) to 192.168.254.4:2060:
  301. OPTIONS sip:400@192.168.254.4:2060;line=hv0dbr9y SIP/2.0
  302. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK30c275f3;rport
  303. Max-Forwards: 70
  304. From: "Unknown" <sip:Unknown@192.168.254.250>;tag=as792aa592
  305. To: <sip:400@192.168.254.4:2060;line=hv0dbr9y>
  306. Contact: <sip:Unknown@192.168.254.250>
  307. Call-ID: 455605116484db2b3aa3219d1102e0b9@192.168.254.250
  308. CSeq: 102 OPTIONS
  309. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  310. Date: Tue, 22 Jun 2010 14:35:57 GMT
  311. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  312. Supported: replaces, timer
  313. Content-Length: 0
  314.  
  315.  
  316. ---
  317.  
  318. trixbox1*CLI>
  319. <--- SIP read from UDP://192.168.254.4:2060 --->
  320. SIP/2.0 200 OK
  321. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK30c275f3;rport=5060
  322. From: "Unknown" <sip:Unknown@192.168.254.250>;tag=as792aa592
  323. To: <sip:400@192.168.254.4:2060;line=hv0dbr9y>
  324. Call-ID: 455605116484db2b3aa3219d1102e0b9@192.168.254.250
  325. CSeq: 102 OPTIONS
  326. Contact: <sip:301@192.168.254.4:2060;line=enbcb1mj>
  327. User-Agent: snom190/3.60x
  328. Accept-Language: en
  329. Accept: application/sdp
  330. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  331. Allow-Events: talk, hold, refer
  332. Supported: timer, 100rel, replaces
  333. Content-Length: 0
  334.  
  335.  
  336. <------------->
  337. --- (14 headers 0 lines) ---
  338. Really destroying SIP dialog '455605116484db2b3aa3219d1102e0b9@192.168.254.250' Method: OPTIONS
  339.  
  340. trixbox1*CLI>
  341. Retransmitting #1 (NAT) to 85.223.99.38:41532:
  342. INVITE sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4 SIP/2.0
  343. Via: SIP/2.0/UDP 195.241.40.168:5060;branch=z9hG4bK51bd45ae;rport
  344. Max-Forwards: 70
  345. From: "MDLanonymous" <sip:private@195.241.40.168>;tag=as3a842f28
  346. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  347. Contact: <sip:private@195.241.40.168>
  348. Call-ID: 1850333112755b7129c2ce8f70a2a644@195.241.40.168
  349. CSeq: 102 INVITE
  350. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  351. Date: Tue, 22 Jun 2010 14:35:57 GMT
  352. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  353. Supported: replaces, timer
  354. Content-Type: application/sdp
  355. Content-Length: 403
  356.  
  357. v=0
  358. o=root 1179129330 1179129330 IN IP4 195.241.40.168
  359. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  360. c=IN IP4 195.241.40.168
  361. b=CT:384
  362. t=0 0
  363. m=audio 11000 RTP/AVP 8 0 101
  364. a=rtpmap:8 PCMA/8000
  365. a=rtpmap:0 PCMU/8000
  366. a=rtpmap:101 telephone-event/8000
  367. a=fmtp:101 0-16
  368. a=silenceSupp:off - - - -
  369. a=ptime:20
  370. a=sendrecv
  371. m=video 12232 RTP/AVP 34 99
  372. a=rtpmap:34 H263/90000
  373. a=rtpmap:99 H264/90000
  374. a=sendrecv
  375.  
  376. ---
  377.  
  378. trixbox1*CLI>
  379. <--- SIP read from UDP://85.223.99.38:41532 --->
  380. SIP/2.0 180 Ringing
  381. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  382. Contact: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  383. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  384. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  385. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  386. CSeq: 102 INVITE
  387. User-Agent: X-Lite release 1002tx stamp 29712
  388. Content-Length: 0
  389.  
  390.  
  391. <------------->
  392. --- (9 headers 0 lines) ---
  393.  
  394. <--- SIP read from UDP://85.223.99.38:41532 --->
  395.  
  396.  
  397.  
  398. <------------->
  399.  
  400. trixbox1*CLI>
  401. Retransmitting #2 (NAT) to 85.223.99.38:41532:
  402. INVITE sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4 SIP/2.0
  403. Via: SIP/2.0/UDP 195.241.40.168:5060;branch=z9hG4bK51bd45ae;rport
  404. Max-Forwards: 70
  405. From: "MDLanonymous" <sip:private@195.241.40.168>;tag=as3a842f28
  406. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  407. Contact: <sip:private@195.241.40.168>
  408. Call-ID: 1850333112755b7129c2ce8f70a2a644@195.241.40.168
  409. CSeq: 102 INVITE
  410. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  411. Date: Tue, 22 Jun 2010 14:35:57 GMT
  412. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  413. Supported: replaces, timer
  414. Content-Type: application/sdp
  415. Content-Length: 403
  416.  
  417. v=0
  418. o=root 1179129330 1179129330 IN IP4 195.241.40.168
  419. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  420. c=IN IP4 195.241.40.168
  421. b=CT:384
  422. t=0 0
  423. m=audio 11000 RTP/AVP 8 0 101
  424. a=rtpmap:8 PCMA/8000
  425. a=rtpmap:0 PCMU/8000
  426. a=rtpmap:101 telephone-event/8000
  427. a=fmtp:101 0-16
  428. a=silenceSupp:off - - - -
  429. a=ptime:20
  430. a=sendrecv
  431. m=video 12232 RTP/AVP 34 99
  432. a=rtpmap:34 H263/90000
  433. a=rtpmap:99 H264/90000
  434. a=sendrecv
  435.  
  436. ---
  437.  
  438. trixbox1*CLI>
  439. <--- SIP read from UDP://85.223.99.38:41532 --->
  440. SIP/2.0 180 Ringing
  441. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  442. Contact: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  443. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  444. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  445. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  446. CSeq: 102 INVITE
  447. User-Agent: X-Lite release 1002tx stamp 29712
  448. Content-Length: 0
  449.  
  450.  
  451. <------------->
  452. --- (9 headers 0 lines) ---
  453.  
  454. trixbox1*CLI>
  455. Really destroying SIP dialog '069d0b0f2b4aee63266af4b55ca57f55@127.0.0.1' Method: REGISTER
  456. Really destroying SIP dialog '79f2adc4667eefda796770c07f13ddc2@127.0.0.1' Method: REGISTER
  457.  
  458. trixbox1*CLI>
  459. Really destroying SIP dialog '0581f2ae5fa3bead6570e86279cf95d7@127.0.0.1' Method: REGISTER
  460.  
  461. trixbox1*CLI>
  462. Retransmitting #3 (NAT) to 85.223.99.38:41532:
  463. INVITE sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4 SIP/2.0
  464. Via: SIP/2.0/UDP 195.241.40.168:5060;branch=z9hG4bK51bd45ae;rport
  465. Max-Forwards: 70
  466. From: "MDLanonymous" <sip:private@195.241.40.168>;tag=as3a842f28
  467. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  468. Contact: <sip:private@195.241.40.168>
  469. Call-ID: 1850333112755b7129c2ce8f70a2a644@195.241.40.168
  470. CSeq: 102 INVITE
  471. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  472. Date: Tue, 22 Jun 2010 14:35:57 GMT
  473. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  474. Supported: replaces, timer
  475. Content-Type: application/sdp
  476. Content-Length: 403
  477.  
  478. v=0
  479. o=root 1179129330 1179129330 IN IP4 195.241.40.168
  480. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  481. c=IN IP4 195.241.40.168
  482. b=CT:384
  483. t=0 0
  484. m=audio 11000 RTP/AVP 8 0 101
  485. a=rtpmap:8 PCMA/8000
  486. a=rtpmap:0 PCMU/8000
  487. a=rtpmap:101 telephone-event/8000
  488. a=fmtp:101 0-16
  489. a=silenceSupp:off - - - -
  490. a=ptime:20
  491. a=sendrecv
  492. m=video 12232 RTP/AVP 34 99
  493. a=rtpmap:34 H263/90000
  494. a=rtpmap:99 H264/90000
  495. a=sendrecv
  496.  
  497. ---
  498.  
  499. trixbox1*CLI>
  500. <--- SIP read from UDP://85.223.99.38:41532 --->
  501. SIP/2.0 180 Ringing
  502. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  503. Contact: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  504. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  505. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  506. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  507. CSeq: 102 INVITE
  508. User-Agent: X-Lite release 1002tx stamp 29712
  509. Content-Length: 0
  510.  
  511.  
  512. <------------->
  513. --- (9 headers 0 lines) ---
  514.  
  515. trixbox1*CLI>
  516. == Manager 'admin' logged on from 127.0.0.1
  517.  
  518. trixbox1*CLI>
  519. == Manager 'admin' logged off from 127.0.0.1
  520.  
  521. trixbox1*CLI>
  522. Retransmitting #4 (NAT) to 85.223.99.38:41532:
  523. INVITE sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4 SIP/2.0
  524. Via: SIP/2.0/UDP 195.241.40.168:5060;branch=z9hG4bK51bd45ae;rport
  525. Max-Forwards: 70
  526. From: "MDLanonymous" <sip:private@195.241.40.168>;tag=as3a842f28
  527. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  528. Contact: <sip:private@195.241.40.168>
  529. Call-ID: 1850333112755b7129c2ce8f70a2a644@195.241.40.168
  530. CSeq: 102 INVITE
  531. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  532. Date: Tue, 22 Jun 2010 14:35:57 GMT
  533. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  534. Supported: replaces, timer
  535. Content-Type: application/sdp
  536. Content-Length: 403
  537.  
  538. v=0
  539. o=root 1179129330 1179129330 IN IP4 195.241.40.168
  540. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  541. c=IN IP4 195.241.40.168
  542. b=CT:384
  543. t=0 0
  544. m=audio 11000 RTP/AVP 8 0 101
  545. a=rtpmap:8 PCMA/8000
  546. a=rtpmap:0 PCMU/8000
  547. a=rtpmap:101 telephone-event/8000
  548. a=fmtp:101 0-16
  549. a=silenceSupp:off - - - -
  550. a=ptime:20
  551. a=sendrecv
  552. m=video 12232 RTP/AVP 34 99
  553. a=rtpmap:34 H263/90000
  554. a=rtpmap:99 H264/90000
  555. a=sendrecv
  556.  
  557. ---
  558.  
  559. trixbox1*CLI>
  560. <--- SIP read from UDP://85.223.99.38:41532 --->
  561. SIP/2.0 180 Ringing
  562. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  563. Contact: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>
  564. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  565. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  566. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  567. CSeq: 102 INVITE
  568. User-Agent: X-Lite release 1002tx stamp 29712
  569. Content-Length: 0
  570.  
  571.  
  572. <------------->
  573. --- (9 headers 0 lines) ---
  574.  
  575. trixbox1*CLI>
  576. <--- SIP read from UDP://85.223.99.38:41532 --->
  577. SIP/2.0 480 Temporarily Unavailable
  578. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  579. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  580. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  581. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  582. CSeq: 102 INVITE
  583. User-Agent: X-Lite release 1002tx stamp 29712
  584. Content-Length: 0
  585.  
  586.  
  587. <------------->
  588. --- (8 headers 0 lines) ---
  589.  
  590. trixbox1*CLI>
  591. <--- SIP read from UDP://85.223.99.38:41532 --->
  592. SIP/2.0 480 Temporarily Unavailable
  593. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  594. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  595. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  596. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  597. CSeq: 102 INVITE
  598. User-Agent: X-Lite release 1002tx stamp 29712
  599. Content-Length: 0
  600.  
  601.  
  602. <------------->
  603. --- (8 headers 0 lines) ---
  604.  
  605. trixbox1*CLI>
  606. == Manager 'admin' logged on from 127.0.0.1
  607.  
  608. trixbox1*CLI>
  609. <--- SIP read from UDP://85.223.99.38:41532 --->
  610. SIP/2.0 480 Temporarily Unavailable
  611. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  612. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  613. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  614. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  615. CSeq: 102 INVITE
  616. User-Agent: X-Lite release 1002tx stamp 29712
  617. Content-Length: 0
  618.  
  619.  
  620. <------------->
  621. --- (8 headers 0 lines) ---
  622.  
  623. trixbox1*CLI>
  624. -- Nobody picked up in 15000 ms
  625. Scheduling destruction of SIP dialog '1850333112755b7129c2ce8f70a2a644@195.241.40.168' in 32000 ms (Method: INVITE)
  626. -- Executing [s@macro-dial:8] Set("SIP/31229707735-0a5a6ca0", "DIALSTATUS=NOANSWER") in new stack
  627. -- Executing [s@macro-dial:9] GosubIf("SIP/31229707735-0a5a6ca0", "0?NOANSWER,1") in new stack
  628. -- Executing [s@macro-exten-vm:10] GotoIf("SIP/31229707735-0a5a6ca0", "0?exit,return") in new stack
  629. -- Executing [s@macro-exten-vm:11] Set("SIP/31229707735-0a5a6ca0", "SV_DIALSTATUS=NOANSWER") in new stack
  630. -- Executing [s@macro-exten-vm:12] GosubIf("SIP/31229707735-0a5a6ca0", "0?docfu,1") in new stack
  631. -- Executing [s@macro-exten-vm:13] GosubIf("SIP/31229707735-0a5a6ca0", "0?docfb,1") in new stack
  632. -- Executing [s@macro-exten-vm:14] Set("SIP/31229707735-0a5a6ca0", "DIALSTATUS=NOANSWER") in new stack
  633. -- Executing [s@macro-exten-vm:15] NoOp("SIP/31229707735-0a5a6ca0", "Voicemail is '200'") in new stack
  634.  
  635. trixbox1*CLI>
  636. -- Executing [s@macro-exten-vm:16] GotoIf("SIP/31229707735-0a5a6ca0", "0?s-NOANSWER,1") in new stack
  637. -- Executing [s@macro-exten-vm:17] NoOp("SIP/31229707735-0a5a6ca0", "Sending to Voicemail box 200") in new stack
  638. -- Executing [s@macro-exten-vm:18] Macro("SIP/31229707735-0a5a6ca0", "vm,200,NOANSWER,") in new stack
  639.  
  640. trixbox1*CLI>
  641. -- Executing [s@macro-vm:1] Macro("SIP/31229707735-0a5a6ca0", "user-callerid,SKIPTTL") in new stack
  642. -- Executing [s@macro-user-callerid:1] Set("SIP/31229707735-0a5a6ca0", "AMPUSER=private") in new stack
  643. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/31229707735-0a5a6ca0", "0?report") in new stack
  644. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/31229707735-0a5a6ca0", "0?Set(REALCALLERIDNUM=private)") in new stack
  645.  
  646. trixbox1*CLI>
  647. -- Executing [s@macro-user-callerid:4] Set("SIP/31229707735-0a5a6ca0", "AMPUSER=") in new stack
  648. -- Executing [s@macro-user-callerid:5] Set("SIP/31229707735-0a5a6ca0", "AMPUSERCIDNAME=") in new stack
  649. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/31229707735-0a5a6ca0", "1?report") in new stack
  650. -- Goto (macro-user-callerid,s,11)
  651. -- Executing [s@macro-user-callerid:11] GotoIf("SIP/31229707735-0a5a6ca0", "1?continue") in new stack
  652. -- Goto (macro-user-callerid,s,20)
  653. -- Executing [s@macro-user-callerid:20] NoOp("SIP/31229707735-0a5a6ca0", "Using CallerID "MDLanonymous" <private>") in new stack
  654. -- Executing [s@macro-vm:2] Set("SIP/31229707735-0a5a6ca0", "VMGAIN=""") in new stack
  655. -- Executing [s@macro-vm:3] GotoIf("SIP/31229707735-0a5a6ca0", "1?vmx,1") in new stack
  656. -- Goto (macro-vm,vmx,1)
  657. -- Executing [vmx@macro-vm:1] GotoIf("SIP/31229707735-0a5a6ca0", "0?s-NOANSWER,1") in new stack
  658. -- Executing [vmx@macro-vm:2] Set("SIP/31229707735-0a5a6ca0", "MODE=unavail") in new stack
  659. -- Executing [vmx@macro-vm:3] GotoIf("SIP/31229707735-0a5a6ca0", "1?notdirect") in new stack
  660. -- Goto (macro-vm,vmx,5)
  661. -- Executing [vmx@macro-vm:5] NoOp("SIP/31229707735-0a5a6ca0", "Checking if ext 200 is enabled: ") in new stack
  662. -- Executing [vmx@macro-vm:6] GotoIf("SIP/31229707735-0a5a6ca0", "1?s-NOANSWER,1") in new stack
  663. -- Goto (macro-vm,s-NOANSWER,1)
  664. -- Executing [s-NOANSWER@macro-vm:1] Macro("SIP/31229707735-0a5a6ca0", "get-vmcontext,200") in new stack
  665. -- Executing [s@macro-get-vmcontext:1] Set("SIP/31229707735-0a5a6ca0", "VMCONTEXT=default") in new stack
  666. -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/31229707735-0a5a6ca0", "0?200:300") in new stack
  667. -- Goto (macro-get-vmcontext,s,300)
  668. -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/31229707735-0a5a6ca0", "") in new stack
  669. -- Executing [s-NOANSWER@macro-vm:2] VoiceMail("SIP/31229707735-0a5a6ca0", "200@default,u""") in new stack
  670. Audio is at 195.241.40.168 port 19326
  671. Adding codec 0x8 (alaw) to SDP
  672. Adding non-codec 0x1 (telephone-event) to SDP
  673.  
  674. <--- Reliably Transmitting (no NAT) to 62.177.135.41:38383 --->
  675. SIP/2.0 200 OK
  676. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK8bcd.d05ed0c1.0;received=62.177.135.41
  677. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK147bacdb
  678. Record-Route: <sip:62.177.135.41:38383;ftag=as443a600e;lr=on>
  679. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  680. To: <sip:31229707736@62.177.135.41:38383>;tag=as66f139a0
  681. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  682. CSeq: 102 INVITE
  683. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  684. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  685. Supported: replaces, timer
  686. Contact: <sip:s@195.241.40.168>
  687. Content-Type: application/sdp
  688. Content-Length: 280
  689.  
  690. v=0
  691. o=root 1331144144 1331144144 IN IP4 195.241.40.168
  692. s=Asterisk PBX 1.6.0.10-FONCORE-r40
  693. c=IN IP4 195.241.40.168
  694. t=0 0
  695. m=audio 19326 RTP/AVP 8 101
  696. a=rtpmap:8 PCMA/8000
  697. a=rtpmap:101 telephone-event/8000
  698. a=fmtp:101 0-16
  699. a=silenceSupp:off - - - -
  700. a=ptime:20
  701. a=sendrecv
  702.  
  703. <------------>
  704.  
  705. trixbox1*CLI>
  706. <--- SIP read from UDP://62.177.135.41:38383 --->
  707. ACK sip:s@195.241.40.168:5060 SIP/2.0
  708. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=0
  709.  
  710. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK5ae54f99
  711. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  712. To: <sip:31229707736@62.177.135.41:38383>;tag=as66f139a0
  713. Contact: <sip:private@62.177.135.40>
  714. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  715. CSeq: 102 ACK
  716. User-Agent: bbned PBX
  717. Max-Forwards: 69
  718. Content-Length: 0
  719.  
  720.  
  721. <------------->
  722. --- (11 headers 0 lines) ---
  723.  
  724. trixbox1*CLI>
  725. == Manager 'admin' logged off from 127.0.0.1
  726.  
  727. trixbox1*CLI>
  728. -- <SIP/31229707735-0a5a6ca0> Playing 'vm-theperson.gsm' (language 'en')
  729.  
  730. trixbox1*CLI>
  731. <--- SIP read from UDP://85.223.99.38:41532 --->
  732. SIP/2.0 480 Temporarily Unavailable
  733. Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK51bd45ae;rport=5060
  734. To: <sip:200@192.168.254.250:5060;rinstance=a6c6343397ca12e4>;tag=0273011c
  735. From: "MDLanonymous"<sip:private@195.241.40.168>;tag=as3a842f28
  736. Call-ID: 1850333112755b7129c2ce8f70a2a644@192.168.254.250
  737. CSeq: 102 INVITE
  738. User-Agent: X-Lite release 1002tx stamp 29712
  739. Content-Length: 0
  740.  
  741.  
  742. <------------->
  743. --- (8 headers 0 lines) ---
  744.  
  745. trixbox1*CLI>
  746. -- <SIP/31229707735-0a5a6ca0> Playing 'digits/2.gsm' (language 'en')
  747.  
  748. trixbox1*CLI>
  749. -- <SIP/31229707735-0a5a6ca0> Playing 'digits/0.gsm' (language 'en')
  750.  
  751. trixbox1*CLI>
  752. -- <SIP/31229707735-0a5a6ca0> Playing 'digits/0.gsm' (language 'en')
  753.  
  754. trixbox1*CLI>
  755. <--- SIP read from UDP://62.177.135.41:38383 --->
  756.  
  757. <------------->
  758.  
  759. trixbox1*CLI>
  760. <--- SIP read from UDP://62.177.135.41:38383 --->
  761.  
  762. <------------->
  763.  
  764. trixbox1*CLI>
  765. <--- SIP read from UDP://62.177.135.41:38383 --->
  766.  
  767. <------------->
  768.  
  769. trixbox1*CLI>
  770. -- <SIP/31229707735-0a5a6ca0> Playing 'vm-isunavail.gsm' (language 'en')
  771.  
  772. trixbox1*CLI>
  773. <--- SIP read from UDP://62.177.135.41:38383 --->
  774. BYE sip:s@195.241.40.168:5060 SIP/2.0
  775. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK9bcd.c74b4d92.0
  776. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK5efba9b1
  777. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  778. To: <sip:31229707736@62.177.135.41:38383>;tag=as66f139a0
  779. Contact: <sip:private@62.177.135.40>
  780. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  781. CSeq: 103 BYE
  782. User-Agent: bbned PBX
  783. Max-Forwards: 69
  784. Content-Length: 0
  785.  
  786.  
  787. <------------->
  788. --- (11 headers 0 lines) ---
  789. Sending to 62.177.135.41 : 38383 (no NAT)
  790.  
  791. <--- Transmitting (no NAT) to 62.177.135.41:38383 --->
  792. SIP/2.0 200 OK
  793. Via: SIP/2.0/UDP 62.177.135.41:38383;branch=z9hG4bK9bcd.c74b4d92.0;received=62.177.135.41
  794. Via: SIP/2.0/UDP 62.177.135.40:5060;branch=z9hG4bK5efba9b1
  795. From: "anonymous" <sip:private@sip.ritstele.com>;tag=as443a600e
  796. To: <sip:31229707736@62.177.135.41:38383>;tag=as66f139a0
  797. Call-ID: 5d240579110ecaa5228230cd6090434f@sip.ritstele.com
  798. CSeq: 103 BYE
  799. User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
  800. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  801. Supported: replaces, timer
  802. Content-Length: 0
  803.  
  804.  
  805. <------------>
  806.  
  807. trixbox1*CLI>
  808. == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/31229707735-0a5a6ca0' in macro 'vm'
  809.  
  810. trixbox1*CLI>
  811. == Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/31229707735-0a5a6ca0' in macro 'exten-vm'
  812. == Spawn extension (ext-local, 200, 1) exited non-zero on 'SIP/31229707735-0a5a6ca0'
  813.  
  814. trixbox1*CLI>
  815. Really destroying SIP dialog '5d240579110ecaa5228230cd6090434f@sip.ritstele.com' Method: BYE
  816.  
  817. trixbox1*CLI>
  818. Disconnected from Asterisk server
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