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  1. jeremiahs-MacBook-Pro:~ jeremiahbooker$ ssh root@110.5.42.156
  2. root@110.5.42.156's password:
  3. Last login: Tue Aug 23 19:52:33 2016 from 183.76.169.117
  4. _____ ____ ______ __
  5. | ___| __ ___ ___| _ \| __ ) \/ /
  6. | |_ | '__/ _ \/ _ \ |_) | _ \\ /
  7. | _|| | | __/ __/ __/| |_) / \
  8. |_| |_| \___|\___|_| |____/_/\_\
  9.  
  10. NOTICE! You have 4 notifications! Please log into the UI to see them!
  11.  
  12. Current Network Configuration
  13. +-----------+-------------------+--------------------------------------+
  14. | Interface | MAC Address | IP Addresses |
  15. +-----------+-------------------+--------------------------------------+
  16. | eth0 | 00:50:43:01:72:99 | 192.168.1.3 |
  17. | | | 2408:210:725:f900:250:43ff:fe01:7299 |
  18. | | | fe80::250:43ff:fe01:7299 |
  19. +-----------+-------------------+--------------------------------------+
  20.  
  21. Please note most tasks should be handled through the GUI.
  22. You can access the GUI by typing one of the above IPs in to your web browser.
  23. For support please visit:
  24. http://www.freepbx.org/support-and-professional-services
  25.  
  26. [root@localhost ~]# asterisk -vrrr
  27. Asterisk 11.23.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  28. Created by Mark Spencer <markster@digium.com>
  29. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  30. This is free software, with components licensed under the GNU General Public
  31. License version 2 and other licenses; you are welcome to redistribute it under
  32. certain conditions. Type 'core show license' for details.
  33. =========================================================================
  34. Connected to Asterisk 11.23.0 currently running on localhost (pid = 13421)
  35. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  36. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  37. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK4e5bc0f4;rport
  38. Max-Forwards: 70
  39. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4c2f6043
  40. To: <sip:200@10.0.1.34:5060>
  41. Contact: <sip:asterisk@110.5.42.156:5060>
  42. Call-ID: 500788582620af2e1154e2300bbc26f3@110.5.42.156:5060
  43. CSeq: 102 OPTIONS
  44. User-Agent: Asterisk PBX 11.23.0
  45. Date: Tue, 23 Aug 2016 11:21:06 GMT
  46. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  47. Supported: replaces, timer
  48. Content-Length: 0
  49.  
  50.  
  51. ---
  52.  
  53. <--- SIP read from UDP:183.76.169.117:40408 --->
  54. SIP/2.0 200 OK
  55. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK4e5bc0f4;rport=5060
  56. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4c2f6043
  57. To: <sip:200@10.0.1.34:5060>;tag=1213563351
  58. Call-ID: 500788582620af2e1154e2300bbc26f3@110.5.42.156:5060
  59. CSeq: 102 OPTIONS
  60. Supported: replaces, path, timer
  61. User-Agent: Grandstream GXP1625 1.0.2.27
  62. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  63. Content-Length: 0
  64.  
  65. <------------->
  66. --- (10 headers 0 lines) ---
  67. Really destroying SIP dialog '500788582620af2e1154e2300bbc26f3@110.5.42.156:5060' Method: OPTIONS
  68. localhost*CLI> core set verbose 10
  69. Console verbose was 1 and is now 10.
  70. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  71. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  72. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK61e93639;rport
  73. Max-Forwards: 70
  74. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5e8384bf
  75. To: <sip:200@10.0.1.34:5060>
  76. Contact: <sip:asterisk@110.5.42.156:5060>
  77. Call-ID: 3832978d5bae4b57689f076a5a5874a4@110.5.42.156:5060
  78. CSeq: 102 OPTIONS
  79. User-Agent: Asterisk PBX 11.23.0
  80. Date: Tue, 23 Aug 2016 11:21:26 GMT
  81. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  82. Supported: replaces, timer
  83. Content-Length: 0
  84.  
  85.  
  86. ---
  87.  
  88. <--- SIP read from UDP:183.76.169.117:40408 --->
  89. SIP/2.0 200 OK
  90. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK61e93639;rport=5060
  91. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5e8384bf
  92. To: <sip:200@10.0.1.34:5060>;tag=1770793685
  93. Call-ID: 3832978d5bae4b57689f076a5a5874a4@110.5.42.156:5060
  94. CSeq: 102 OPTIONS
  95. Supported: replaces, path, timer
  96. User-Agent: Grandstream GXP1625 1.0.2.27
  97. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  98. Content-Length: 0
  99.  
  100. <------------->
  101. --- (10 headers 0 lines) ---
  102. Really destroying SIP dialog '3832978d5bae4b57689f076a5a5874a4@110.5.42.156:5060' Method: OPTIONS
  103. [2016-08-23 20:21:37] NOTICE[12706]: manager.c:2640 authenticate: 108.165.2.22 tried to authenticate with nonexistent user 'asmanager'
  104. [2016-08-23 20:21:37] NOTICE[12706]: manager.c:2677 authenticate: 108.165.2.22 failed to authenticate as 'asmanager'
  105.  
  106. <--- SIP read from UDP:183.76.169.117:40408 --->
  107. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  108. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1363186762;rport
  109. From: <sip:200@110.5.42.156>;tag=1744328604
  110. To: <sip:09016192354@110.5.42.156>
  111. Call-ID: 879533400-5060-23@BA.A.B.DE
  112. CSeq: 220 INVITE
  113. Contact: <sip:200@10.0.1.34:5060>
  114. Max-Forwards: 70
  115. User-Agent: Grandstream GXP1625 1.0.2.27
  116. Privacy: none
  117. P-Preferred-Identity: <sip:200@110.5.42.156>
  118. Supported: replaces, path, timer
  119. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  120. Content-Type: application/sdp
  121. Accept: application/sdp, application/dtmf-relay
  122. Content-Length: 326
  123.  
  124. v=0
  125. o=200 8000 8000 IN IP4 10.0.1.34
  126. s=SIP Call
  127. c=IN IP4 10.0.1.34
  128. t=0 0
  129. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  130. a=sendrecv
  131. a=rtpmap:0 PCMU/8000
  132. a=ptime:20
  133. a=rtpmap:8 PCMA/8000
  134. a=rtpmap:18 G729/8000
  135. a=fmtp:18 annexb=no
  136. a=rtpmap:9 G722/8000
  137. a=rtpmap:2 G726-32/8000
  138. a=rtpmap:101 telephone-event/8000
  139. a=fmtp:101 0-15
  140. <------------->
  141. --- (16 headers 16 lines) ---
  142. Sending to 183.76.169.117:40408 (NAT)
  143. Sending to 183.76.169.117:40408 (NAT)
  144. Using INVITE request as basis request - 879533400-5060-23@BA.A.B.DE
  145. Found peer '200' for '200' from 183.76.169.117:40408
  146.  
  147. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  148. SIP/2.0 401 Unauthorized
  149. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1363186762;received=183.76.169.117;rport=40408
  150. From: <sip:200@110.5.42.156>;tag=1744328604
  151. To: <sip:09016192354@110.5.42.156>;tag=as34e84a6b
  152. Call-ID: 879533400-5060-23@BA.A.B.DE
  153. CSeq: 220 INVITE
  154. Server: Asterisk PBX 11.23.0
  155. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  156. Supported: replaces, timer
  157. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b7c5374"
  158. Content-Length: 0
  159.  
  160.  
  161. <------------>
  162. Scheduling destruction of SIP dialog '879533400-5060-23@BA.A.B.DE' in 6400 ms (Method: INVITE)
  163.  
  164. <--- SIP read from UDP:183.76.169.117:40408 --->
  165. ACK sip:09016192354@110.5.42.156 SIP/2.0
  166. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1363186762;rport
  167. From: <sip:200@110.5.42.156>;tag=1744328604
  168. To: <sip:09016192354@110.5.42.156>;tag=as34e84a6b
  169. Call-ID: 879533400-5060-23@BA.A.B.DE
  170. CSeq: 220 ACK
  171. Content-Length: 0
  172.  
  173. <------------->
  174. --- (7 headers 0 lines) ---
  175.  
  176. <--- SIP read from UDP:183.76.169.117:40408 --->
  177. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  178. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1253510292;rport
  179. From: <sip:200@110.5.42.156>;tag=1744328604
  180. To: <sip:09016192354@110.5.42.156>
  181. Call-ID: 879533400-5060-23@BA.A.B.DE
  182. CSeq: 221 INVITE
  183. Contact: <sip:200@10.0.1.34:5060>
  184. Authorization: Digest username="200", realm="asterisk", nonce="6b7c5374", uri="sip:09016192354@110.5.42.156", response="1e7359a0f24ebda62b3d62a75f85d524", algorithm=MD5
  185. Max-Forwards: 70
  186. User-Agent: Grandstream GXP1625 1.0.2.27
  187. Privacy: none
  188. P-Preferred-Identity: <sip:200@110.5.42.156>
  189. Supported: replaces, path, timer
  190. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  191. Content-Type: application/sdp
  192. Accept: application/sdp, application/dtmf-relay
  193. Content-Length: 326
  194.  
  195. v=0
  196. o=200 8000 8000 IN IP4 10.0.1.34
  197. s=SIP Call
  198. c=IN IP4 10.0.1.34
  199. t=0 0
  200. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  201. a=sendrecv
  202. a=rtpmap:0 PCMU/8000
  203. a=ptime:20
  204. a=rtpmap:8 PCMA/8000
  205. a=rtpmap:18 G729/8000
  206. a=fmtp:18 annexb=no
  207. a=rtpmap:9 G722/8000
  208. a=rtpmap:2 G726-32/8000
  209. a=rtpmap:101 telephone-event/8000
  210. a=fmtp:101 0-15
  211. <------------->
  212. --- (17 headers 16 lines) ---
  213. Sending to 183.76.169.117:40408 (NAT)
  214. Using INVITE request as basis request - 879533400-5060-23@BA.A.B.DE
  215. Found peer '200' for '200' from 183.76.169.117:40408
  216. == Using SIP RTP CoS mark 5
  217. Found RTP audio format 0
  218. Found RTP audio format 8
  219. Found RTP audio format 18
  220. Found RTP audio format 9
  221. Found RTP audio format 2
  222. Found RTP audio format 101
  223. Found audio description format PCMU for ID 0
  224. Found audio description format PCMA for ID 8
  225. Found audio description format G729 for ID 18
  226. Found audio description format G722 for ID 9
  227. Found audio description format G726-32 for ID 2
  228. Found audio description format telephone-event for ID 101
  229. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  230. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  231. Peer audio RTP is at port 10.0.1.34:5004
  232. Looking for 09016192354 in from-internal (domain 110.5.42.156)
  233. list_route: hop: <sip:200@10.0.1.34:5060>
  234.  
  235. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  236. SIP/2.0 100 Trying
  237. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1253510292;received=183.76.169.117;rport=40408
  238. From: <sip:200@110.5.42.156>;tag=1744328604
  239. To: <sip:09016192354@110.5.42.156>
  240. Call-ID: 879533400-5060-23@BA.A.B.DE
  241. CSeq: 221 INVITE
  242. Server: Asterisk PBX 11.23.0
  243. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  244. Supported: replaces, timer
  245. Session-Expires: 1800;refresher=uas
  246. Contact: <sip:09016192354@110.5.42.156:5060>
  247. Content-Length: 0
  248.  
  249.  
  250. <------------>
  251. -- Executing [09016192354@from-internal:1] Macro("SIP/200-000002b8", "user-callerid,LIMIT,EXTERNAL,") in new stack
  252. -- Executing [s@macro-user-callerid:1] Set("SIP/200-000002b8", "TOUCH_MONITOR=1471951301.696") in new stack
  253. -- Executing [s@macro-user-callerid:2] Set("SIP/200-000002b8", "AMPUSER=200") in new stack
  254. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/200-000002b8", "0?report") in new stack
  255. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/200-000002b8", "1?Set(REALCALLERIDNUM=200)") in new stack
  256. -- Executing [s@macro-user-callerid:5] Set("SIP/200-000002b8", "AMPUSER=200") in new stack
  257. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/200-000002b8", "0?limit") in new stack
  258. -- Executing [s@macro-user-callerid:7] Set("SIP/200-000002b8", "AMPUSERCIDNAME=Helpdesk") in new stack
  259. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/200-000002b8", "0?report") in new stack
  260. -- Executing [s@macro-user-callerid:9] Set("SIP/200-000002b8", "AMPUSERCID=200") in new stack
  261. -- Executing [s@macro-user-callerid:10] Set("SIP/200-000002b8", "__DIAL_OPTIONS=Ttr") in new stack
  262. -- Executing [s@macro-user-callerid:11] Set("SIP/200-000002b8", "CALLERID(all)="Helpdesk" <200>") in new stack
  263. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/200-000002b8", "0?limit") in new stack
  264. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/200-000002b8", "1?Set(GROUP(concurrency_limit)=200)") in new stack
  265. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/200-000002b8", "1?Set(CHANNEL(language)=ja)") in new stack
  266. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/200-000002b8", "1?continue") in new stack
  267. -- Goto (macro-user-callerid,s,29)
  268. -- Executing [s@macro-user-callerid:29] Set("SIP/200-000002b8", "CALLERID(number)=200") in new stack
  269. -- Executing [s@macro-user-callerid:30] Set("SIP/200-000002b8", "CALLERID(name)=Helpdesk") in new stack
  270. -- Executing [s@macro-user-callerid:31] Set("SIP/200-000002b8", "CDR(cnum)=200") in new stack
  271. -- Executing [s@macro-user-callerid:32] Set("SIP/200-000002b8", "CDR(cnam)=Helpdesk") in new stack
  272. -- Executing [s@macro-user-callerid:33] Set("SIP/200-000002b8", "CHANNEL(language)=ja") in new stack
  273. -- Executing [09016192354@from-internal:2] NoCDR("SIP/200-000002b8", "") in new stack
  274. -- Executing [09016192354@from-internal:3] Progress("SIP/200-000002b8", "") in new stack
  275. Audio is at 13500
  276. Adding codec 100004 (alaw) to SDP
  277. Adding codec 100003 (ulaw) to SDP
  278. Adding non-codec 0x1 (telephone-event) to SDP
  279.  
  280. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  281. SIP/2.0 183 Session Progress
  282. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1253510292;received=183.76.169.117;rport=40408
  283. From: <sip:200@110.5.42.156>;tag=1744328604
  284. To: <sip:09016192354@110.5.42.156>;tag=as1f068318
  285. Call-ID: 879533400-5060-23@BA.A.B.DE
  286. CSeq: 221 INVITE
  287. Server: Asterisk PBX 11.23.0
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  289. Supported: replaces, timer
  290. Session-Expires: 1800;refresher=uas
  291. Contact: <sip:09016192354@110.5.42.156:5060>
  292. Content-Type: application/sdp
  293. Require: timer
  294. Content-Length: 260
  295.  
  296. v=0
  297. o=root 1461642043 1461642043 IN IP4 110.5.42.156
  298. s=Asterisk PBX 11.23.0
  299. c=IN IP4 110.5.42.156
  300. t=0 0
  301. m=audio 13500 RTP/AVP 8 0 101
  302. a=rtpmap:8 PCMA/8000
  303. a=rtpmap:0 PCMU/8000
  304. a=rtpmap:101 telephone-event/8000
  305. a=fmtp:101 0-16
  306. a=ptime:20
  307. a=sendrecv
  308.  
  309. <------------>
  310. -- Executing [09016192354@from-internal:4] Wait("SIP/200-000002b8", "1") in new stack
  311. > 0x7f2edc2405b0 -- Probation passed - setting RTP source address to 183.76.169.117:41997
  312. > 0x7f2edc2405b0 -- Probation passed - setting RTP source address to 183.76.169.117:41997
  313. -- Executing [09016192354@from-internal:5] Playback("SIP/200-000002b8", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
  314. -- <SIP/200-000002b8> Playing 'silence/1.alaw' (language 'ja')
  315. -- <SIP/200-000002b8> Playing 'cannot-complete-as-dialed.alaw' (language 'ja')
  316. -- <SIP/200-000002b8> Playing 'check-number-dial-again.alaw' (language 'ja')
  317. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  318. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  319. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK13515f07;rport
  320. Max-Forwards: 70
  321. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as6c7043da
  322. To: <sip:200@10.0.1.34:5060>
  323. Contact: <sip:asterisk@110.5.42.156:5060>
  324. Call-ID: 1b201b9d527dde710f723bb95bb4553a@110.5.42.156:5060
  325. CSeq: 102 OPTIONS
  326. User-Agent: Asterisk PBX 11.23.0
  327. Date: Tue, 23 Aug 2016 11:21:46 GMT
  328. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  329. Supported: replaces, timer
  330. Content-Length: 0
  331.  
  332.  
  333. ---
  334.  
  335. <--- SIP read from UDP:183.76.169.117:40408 --->
  336. SIP/2.0 200 OK
  337. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK13515f07;rport=5060
  338. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as6c7043da
  339. To: <sip:200@10.0.1.34:5060>;tag=1944121522
  340. Call-ID: 1b201b9d527dde710f723bb95bb4553a@110.5.42.156:5060
  341. CSeq: 102 OPTIONS
  342. Supported: replaces, path, timer
  343. User-Agent: Grandstream GXP1625 1.0.2.27
  344. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  345. Content-Length: 0
  346.  
  347. <------------->
  348. --- (10 headers 0 lines) ---
  349. Really destroying SIP dialog '1b201b9d527dde710f723bb95bb4553a@110.5.42.156:5060' Method: OPTIONS
  350. -- Executing [09016192354@from-internal:6] Wait("SIP/200-000002b8", "1") in new stack
  351. -- Executing [09016192354@from-internal:7] Congestion("SIP/200-000002b8", "20") in new stack
  352.  
  353. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  354. SIP/2.0 503 Service Unavailable
  355. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1253510292;received=183.76.169.117;rport=40408
  356. From: <sip:200@110.5.42.156>;tag=1744328604
  357. To: <sip:09016192354@110.5.42.156>;tag=as1f068318
  358. Call-ID: 879533400-5060-23@BA.A.B.DE
  359. CSeq: 221 INVITE
  360. Server: Asterisk PBX 11.23.0
  361. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  362. Supported: replaces, timer
  363. Session-Expires: 1800;refresher=uas
  364. Content-Length: 0
  365.  
  366.  
  367. <------------>
  368. [2016-08-23 20:21:50] WARNING[12718][C-000002c7]: channel.c:4861 ast_prod: Prodding channel 'SIP/200-000002b8' failed
  369. == Spawn extension (from-internal, 09016192354, 7) exited non-zero on 'SIP/200-000002b8'
  370. -- Executing [h@from-internal:1] Macro("SIP/200-000002b8", "hangupcall") in new stack
  371. -- Executing [s@macro-hangupcall:1] ExecIf("SIP/200-000002b8", "0?Set(CDR(recordingfile)=.)") in new stack
  372. -- Executing [s@macro-hangupcall:2] GotoIf("SIP/200-000002b8", "1?theend") in new stack
  373. -- Goto (macro-hangupcall,s,4)
  374. -- Executing [s@macro-hangupcall:4] ExecIf("SIP/200-000002b8", "0?Set(CDR(recordingfile)=)") in new stack
  375. -- Executing [s@macro-hangupcall:5] Hangup("SIP/200-000002b8", "") in new stack
  376. == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/200-000002b8' in macro 'hangupcall'
  377. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-000002b8'
  378.  
  379. <--- SIP read from UDP:183.76.169.117:40408 --->
  380. ACK sip:09016192354@110.5.42.156 SIP/2.0
  381. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1253510292;rport
  382. From: <sip:200@110.5.42.156>;tag=1744328604
  383. To: <sip:09016192354@110.5.42.156>;tag=as1f068318
  384. Call-ID: 879533400-5060-23@BA.A.B.DE
  385. CSeq: 221 ACK
  386. Content-Length: 0
  387.  
  388. <------------->
  389. --- (7 headers 0 lines) ---
  390. Really destroying SIP dialog '879533400-5060-23@BA.A.B.DE' Method: ACK
  391. localhost*CLI>
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