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  1. Parsing /etc/asterisk/asterisk.conf
  2. Seeding global EID '00:0c:29:94:22:7d' from 'eth0' using 'siocgifhwaddr'
  3. Asterisk 11.3.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  4. Created by Mark Spencer <markster@digium.com>
  5. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  6. This is free software, with components licensed under the GNU General Public
  7. License version 2 and other licenses; you are welcome to redistribute it under
  8. certain conditions. Type 'core show license' for details.
  9. =========================================================================
  10. Connected to Asterisk 11.3.0 currently running on asteros (pid = 2469)
  11. asteros*CLI> quitreload[4@sip rset debug onreloadset debug on
  12. asteros*CLI>
  13. SIP Debugging re-enabled
  14.  
  15. asteros*CLI> sip set debug onquitreload[4@sip rr[4@sip r
  16. asteros*CLI>
  17.  Reloading SIP
  18.  
  19. asteros*CLI> rel
  20. 
  21. <--- SIP read from UDP:192.168.127.102:4134 --->
  22.  
  23.  
  24. <------------->
  25.  
  26. asteros*CLI> reload
  27. asteros*CLI>
  28.  == Parsing '/etc/asterisk/extconfig.conf': Found
  29. == Parsing '/etc/asterisk/logger.conf': Found
  30. Asterisk Queue Logger restarted
  31. == Parsing '/etc/asterisk/cel.conf': Found
  32. -- CEL logging disabled.
  33. == Parsing '/etc/asterisk/codecs.conf': Found
  34. -- Reloading module 'app_amd.so' (Answering Machine Detection Application)
  35. -- Reloading module 'app_confbridge.so' (Conference Bridge Application)
  36. -- Reloading module 'app_followme.so' (Find-Me/Follow-Me Application)
  37. -- Reloading module 'app_minivm.so' (Mini VoiceMail (A minimal Voicemail e-mail System))
  38. -- Reloading module 'app_playback.so' (Sound File Playback Application)
  39. -- Reloading module 'app_queue.so' (True Call Queueing)
  40. [May 7 13:04:25] NOTICE[3987]: app_queue.c:7712 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
  41. -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
  42. -- Reloading module 'cdr_adaptive_odbc.so' (Adaptive ODBC CDR backend)
  43. == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': Found
  44. -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
  45. -- Reloading module 'cdr_custom.so' (Customizable Comma Separated Values CDR Backend)
  46. == Parsing '/etc/asterisk/cdr_custom.conf': Found
  47. -- Reloading module 'cdr_manager.so' (Asterisk Manager Interface CDR Backend)
  48. -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend)
  49. -- Reloading module 'cel_custom.so' (Customizable Comma Separated Values CEL Backend)
  50. == Parsing '/etc/asterisk/cel_custom.conf': Found
  51. -- Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend)
  52. -- Reloading module 'cel_odbc.so' (ODBC CEL backend)
  53. == Parsing '/etc/asterisk/cel_odbc.conf': Found
  54. -- Reloading module 'chan_agent.so' (Agent Proxy Channel)
  55. -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
  56. -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
  57. Reloading SIP
  58. -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny))
  59. [May 7 13:04:25] NOTICE[3987]: chan_skinny.c:7732 config_load: Configuring skinny from skinny.conf
  60. == Parsing '/etc/asterisk/skinny.conf': Found
  61. -- Reloading module 'chan_unistim.so' (UNISTIM Protocol (USTM))
  62. Reloading unistim.conf...
  63. == Parsing '/etc/asterisk/unistim.conf': Found
  64. -- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decoder)
  65. -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
  66. -- Reloading module 'codec_g722.so' (ITU G.722-64kbps G722 Transcoder)
  67. -- Reloading module 'codec_g726.so' (ITU G.726-32kbps G726 Transcoder)
  68. -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder)
  69. -- Reloading module 'codec_lpc10.so' (LPC10 2.4kbps Coder/Decoder)
  70. -- Reloading module 'codec_ulaw.so' (mu-Law Coder/Decoder)
  71. -- Reloading module 'func_odbc.so' (ODBC lookups)
  72. -- Reloading module 'pbx_ael.so' (Asterisk Extension Language Compiler)
  73. [May 7 13:04:25] NOTICE[3987]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
  74. [May 7 13:04:25] NOTICE[3987]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
  75. [May 7 13:04:25] NOTICE[3987]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
  76. == Setting global variable 'CONSOLE-AEL' to '"Console/dsp"'
  77. == Setting global variable 'IAXINFO-AEL' to 'guest'
  78. == Setting global variable 'OUTBOUND-TRUNK' to '"Zap/g2"'
  79. == Setting global variable 'OUTBOUND-TRUNKMSD' to '1'
  80. -- Registered extension context 'ael-dundi-e164-canonical'; registrar: pbx_ael
  81. -- Registered extension context 'ael-dundi-e164-customers'; registrar: pbx_ael
  82. -- Registered extension context 'ael-dundi-e164-via-pstn'; registrar: pbx_ael
  83. -- Registered extension context 'ael-dundi-e164-local'; registrar: pbx_ael
  84. -- Including context 'ael-dundi-e164-canonical' in context 'ael-dundi-e164-local'
  85. -- Including context 'ael-dundi-e164-customers' in context 'ael-dundi-e164-local'
  86. -- Including context 'ael-dundi-e164-via-pstn' in context 'ael-dundi-e164-local'
  87. -- Registered extension context 'ael-dundi-e164-switch'; registrar: pbx_ael
  88. -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
  89. -- Registered extension context 'ael-dundi-e164-lookup'; registrar: pbx_ael
  90. -- Including context 'ael-dundi-e164-local' in context 'ael-dundi-e164-lookup'
  91. -- Including context 'ael-dundi-e164-switch' in context 'ael-dundi-e164-lookup'
  92. -- Registered extension context 'ael-dundi-e164'; registrar: pbx_ael
  93. -- Registered extension context 'ael-iaxtel700'; registrar: pbx_ael
  94. -- Registered extension context 'ael-iaxprovider'; registrar: pbx_ael
  95. -- Registered extension context 'ael-trunkint'; registrar: pbx_ael
  96. -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkint'
  97. -- Registered extension context 'ael-trunkld'; registrar: pbx_ael
  98. -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkld'
  99. -- Registered extension context 'ael-trunklocal'; registrar: pbx_ael
  100. -- Registered extension context 'ael-trunktollfree'; registrar: pbx_ael
  101. -- Registered extension context 'ael-international'; registrar: pbx_ael
  102. -- Including context 'ael-longdistance' in context 'ael-international'
  103. -- Including context 'ael-trunkint' in context 'ael-international'
  104. -- Registered extension context 'ael-longdistance'; registrar: pbx_ael
  105. -- Including context 'ael-local' in context 'ael-longdistance'
  106. -- Including context 'ael-trunkld' in context 'ael-longdistance'
  107. -- Registered extension context 'ael-local'; registrar: pbx_ael
  108. -- Including context 'ael-default' in context 'ael-local'
  109. -- Including context 'ael-trunklocal' in context 'ael-local'
  110. -- Including context 'ael-iaxtel700' in context 'ael-local'
  111. -- Including context 'ael-trunktollfree' in context 'ael-local'
  112. -- Including context 'ael-iaxprovider' in context 'ael-local'
  113. -- Registered extension context 'ael-std-exten-ael'; registrar: pbx_ael
  114. -- Registered extension context 'ael-demo'; registrar: pbx_ael
  115. -- Registered extension context 'ael-default'; registrar: pbx_ael
  116. -- Including context 'ael-demo' in context 'ael-default'
  117. -- Registered extension context 'ael-builtin-h-bubble'; registrar: pbx_ael
  118. -- Including context 'ael-builtin-h-bubble' in context 'ael-dundi-e164'
  119. -- Including context 'ael-builtin-h-bubble' in context 'ael-std-exten-ael'
  120. -- Added extension '~~s~~' priority 1 to ael-dundi-e164
  121. -- Added extension '~~s~~' priority 2 to ael-dundi-e164
  122. -- Added extension '~~s~~' priority 3 to ael-dundi-e164
  123. -- Added extension '_91700XXXXXXX' priority 1 to ael-iaxtel700
  124. -- Added extension '_9011.' priority 1 to ael-trunkint
  125. -- Added extension '_9011.' priority 2 to ael-trunkint
  126. -- Added extension '_91NXXNXXXXXX' priority 1 to ael-trunkld
  127. -- Added extension '_91NXXNXXXXXX' priority 2 to ael-trunkld
  128. -- Added extension '_9NXXXXXX' priority 1 to ael-trunklocal
  129. -- Added extension '_91800NXXXXXX' priority 1 to ael-trunktollfree
  130. -- Added extension '_91888NXXXXXX' priority 1 to ael-trunktollfree
  131. -- Added extension '_91877NXXXXXX' priority 1 to ael-trunktollfree
  132. -- Added extension '_91866NXXXXXX' priority 1 to ael-trunktollfree
  133. -- Added extension '~~s~~' priority 1 to ael-std-exten-ael
  134. -- Added extension '~~s~~' priority 2 to ael-std-exten-ael
  135. -- Added extension '~~s~~' priority 3 to ael-std-exten-ael
  136. -- Added extension '~~s~~' priority 4 to ael-std-exten-ael
  137. -- Added extension '~~s~~' priority 5 to ael-std-exten-ael
  138. -- Added extension '~~s~~' priority 6 to ael-std-exten-ael
  139. -- Added extension '~~s~~' priority 7 to ael-std-exten-ael
  140. -- Added extension '~~s~~' priority 8 to ael-std-exten-ael
  141. -- Added extension 'a' priority 1 to ael-std-exten-ael
  142. -- Added extension 'a' priority 2 to ael-std-exten-ael
  143. -- Added extension '_sw_19_.' priority 10 to ael-std-exten-ael
  144. -- Added extension '_sw_19_.' priority 11 to ael-std-exten-ael
  145. -- Added extension 'sw_19_' priority 10 to ael-std-exten-ael
  146. -- Added extension 'sw_19_BUSY' priority 10 to ael-std-exten-ael
  147. -- Added extension 'sw_19_BUSY' priority 11 to ael-std-exten-ael
  148. -- Added extension 's' priority 1 to ael-demo
  149. -- Added extension 's' priority 2 to ael-demo
  150. -- Added extension 's' priority 3 to ael-demo
  151. -- Added extension 's' priority 4 to ael-demo
  152. -- Added extension 's' priority 5 to ael-demo
  153. -- Added extension 's' priority 6 to ael-demo
  154. -- Added extension 's' priority 7 to ael-demo
  155. -- Added extension 's' priority 8 to ael-demo
  156. -- Added extension 's' priority 9 to ael-demo
  157. -- Added extension 's' priority 10 to ael-demo
  158. -- Added extension 's' priority 11 to ael-demo
  159. -- Added extension 's' priority 12 to ael-demo
  160. -- Added extension '2' priority 1 to ael-demo
  161. -- Added extension '2' priority 2 to ael-demo
  162. -- Added extension '3' priority 1 to ael-demo
  163. -- Added extension '3' priority 2 to ael-demo
  164. -- Added extension '1000' priority 1 to ael-demo
  165. -- Added extension '500' priority 1 to ael-demo
  166. -- Added extension '500' priority 2 to ael-demo
  167. -- Added extension '500' priority 3 to ael-demo
  168. -- Added extension '500' priority 4 to ael-demo
  169. -- Added extension '600' priority 1 to ael-demo
  170. -- Added extension '600' priority 2 to ael-demo
  171. -- Added extension '600' priority 3 to ael-demo
  172. -- Added extension '600' priority 4 to ael-demo
  173. -- Added extension '_1234' priority 1 to ael-demo
  174. -- Added extension '8500' priority 1 to ael-demo
  175. -- Added extension '8500' priority 2 to ael-demo
  176. -- Added extension '#' priority 1 to ael-demo
  177. -- Added extension '#' priority 2 to ael-demo
  178. -- Added extension 't' priority 1 to ael-demo
  179. -- Added extension 'i' priority 1 to ael-demo
  180. -- Added extension 'h' priority 1 to ael-builtin-h-bubble
  181. -- Added extension 'h' priority 9991 to ael-builtin-h-bubble
  182. -- Added extension 'h' priority 9992 to ael-builtin-h-bubble
  183. -- Added extension 'h' priority 9993 to ael-builtin-h-bubble
  184. -- Added extension 'h' priority 9994 to ael-builtin-h-bubble
  185. -- Added extension 'h' priority 9995 to ael-builtin-h-bubble
  186. -- Added extension 'h' priority 9996 to ael-builtin-h-bubble
  187. [May 7 13:04:25] NOTICE[3987]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
  188. -- Registered extension context 'parkedcalls'; registrar: features
  189. -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_ael
  190. -- Added extension '700' priority 1 to parkedcalls
  191. -- Registered extension context 'internal'; registrar: pbx_config
  192. -- merging incls/swits/igpats from old(internal) to new(internal) context, registrar = pbx_ael
  193. -- Added extension '*98' priority 4 to internal
  194. -- Added extension '*98' priority 3 to internal
  195. -- Added extension '*98' priority 2 to internal
  196. -- Added extension '*98' priority 1 to internal
  197. -- Added extension '_xxx' priority 4 to internal
  198. -- Added extension '_xxx' priority 3 to internal
  199. -- Added extension '_xxx' priority 2 to internal
  200. -- Added extension '_xxx' priority 1 to internal
  201. -- Registered extension context 'macro-q_logout'; registrar: pbx_config
  202. -- merging incls/swits/igpats from old(macro-q_logout) to new(macro-q_logout) context, registrar = pbx_ael
  203. -- Added extension 's' priority 4 to macro-q_logout
  204. -- Added extension 's' priority 3 to macro-q_logout
  205. -- Added extension 's' priority 2 to macro-q_logout
  206. -- Added extension 's' priority 1 to macro-q_logout
  207. -- Registered extension context 'macro-q_login'; registrar: pbx_config
  208. -- merging incls/swits/igpats from old(macro-q_login) to new(macro-q_login) context, registrar = pbx_ael
  209. -- Added extension 's' priority 4 to macro-q_login
  210. -- Added extension 's' priority 3 to macro-q_login
  211. -- Added extension 's' priority 2 to macro-q_login
  212. -- Added extension 's' priority 1 to macro-q_login
  213. -- Registered extension context 'macro-member-loginlogout'; registrar: pbx_config
  214. -- merging incls/swits/igpats from old(macro-member-loginlogout) to new(macro-member-loginlogout) context, registrar = pbx_ael
  215. -- Added extension 's' priority 14 to macro-member-loginlogout
  216. -- Added extension 's' priority 13 to macro-member-loginlogout
  217. -- Added extension 's' priority 12 to macro-member-loginlogout
  218. -- Added extension 's' priority 11 to macro-member-loginlogout
  219. -- Added extension 's' priority 10 to macro-member-loginlogout
  220. -- Added extension 's' priority 9 to macro-member-loginlogout
  221. -- Added extension 's' priority 8 to macro-member-loginlogout
  222. -- Added extension 's' priority 7 to macro-member-loginlogout
  223. -- Added extension 's' priority 6 to macro-member-loginlogout
  224. -- Added extension 's' priority 5 to macro-member-loginlogout
  225. -- Added extension 's' priority 4 to macro-member-loginlogout
  226. -- Added extension 's' priority 3 to macro-member-loginlogout
  227. -- Added extension 's' priority 2 to macro-member-loginlogout
  228. -- Added extension 's' priority 1 to macro-member-loginlogout
  229. -- Registered extension context 'queue-member-manager'; registrar: pbx_config
  230. -- merging incls/swits/igpats from old(queue-member-manager) to new(queue-member-manager) context, registrar = pbx_ael
  231. -- Added extension 'handle_member' priority 9 to queue-member-manager
  232. -- Added extension 'handle_member' priority 8 to queue-member-manager
  233. -- Added extension 'handle_member' priority 7 to queue-member-manager
  234. -- Added extension 'handle_member' priority 6 to queue-member-manager
  235. -- Added extension 'handle_member' priority 5 to queue-member-manager
  236. -- Added extension 'handle_member' priority 4 to queue-member-manager
  237. -- Added extension 'handle_member' priority 3 to queue-member-manager
  238. -- Added extension 'handle_member' priority 2 to queue-member-manager
  239. -- Added extension 'handle_member' priority 1 to queue-member-manager
  240. -- Registered extension context 'macro-trunkdial-failover-0.3'; registrar: pbx_config
  241. -- merging incls/swits/igpats from old(macro-trunkdial-failover-0.3) to new(macro-trunkdial-failover-0.3) context, registrar = pbx_ael
  242. -- Added extension '1-out' priority 1 to macro-trunkdial-failover-0.3
  243. -- Added extension '1-CONGESTION' priority 2 to macro-trunkdial-failover-0.3
  244. -- Added extension '1-CONGESTION' priority 1 to macro-trunkdial-failover-0.3
  245. -- Added extension '1-CHANUNAVAIL' priority 2 to macro-trunkdial-failover-0.3
  246. -- Added extension '1-CHANUNAVAIL' priority 1 to macro-trunkdial-failover-0.3
  247. -- Added extension '1-dial' priority 2 to macro-trunkdial-failover-0.3
  248. -- Added extension '1-dial' priority 1 to macro-trunkdial-failover-0.3
  249. -- Added extension '1-fmsetcid' priority 3 to macro-trunkdial-failover-0.3
  250. -- Added extension '1-fmsetcid' priority 2 to macro-trunkdial-failover-0.3
  251. -- Added extension '1-fmsetcid' priority 1 to macro-trunkdial-failover-0.3
  252. -- Added extension '1-setgbobname' priority 2 to macro-trunkdial-failover-0.3
  253. -- Added extension '1-setgbobname' priority 1 to macro-trunkdial-failover-0.3
  254. -- Added extension 's' priority 8 to macro-trunkdial-failover-0.3
  255. -- Added extension 's' priority 7 to macro-trunkdial-failover-0.3
  256. -- Added extension 's' priority 6 to macro-trunkdial-failover-0.3
  257. -- Added extension 's' priority 5 to macro-trunkdial-failover-0.3
  258. -- Added extension 's' priority 4 to macro-trunkdial-failover-0.3
  259. -- Added extension 's' priority 3 to macro-trunkdial-failover-0.3
  260. -- Added extension 's' priority 2 to macro-trunkdial-failover-0.3
  261. -- Added extension 's' priority 1 to macro-trunkdial-failover-0.3
  262. -- Registered extension context 'macro-local-callingrule-cid-0.1'; registrar: pbx_config
  263. -- merging incls/swits/igpats from old(macro-local-callingrule-cid-0.1) to new(macro-local-callingrule-cid-0.1) context, registrar = pbx_ael
  264. -- Added extension 's' priority 2 to macro-local-callingrule-cid-0.1
  265. -- Added extension 's' priority 1 to macro-local-callingrule-cid-0.1
  266. -- Registered extension context 'asterisk_guitools'; registrar: pbx_config
  267. -- merging incls/swits/igpats from old(asterisk_guitools) to new(asterisk_guitools) context, registrar = pbx_ael
  268. -- Added extension 'play_file' priority 3 to asterisk_guitools
  269. -- Added extension 'play_file' priority 2 to asterisk_guitools
  270. -- Added extension 'play_file' priority 1 to asterisk_guitools
  271. -- Added extension 'record_vmenu' priority 6 to asterisk_guitools
  272. -- Added extension 'record_vmenu' priority 5 to asterisk_guitools
  273. -- Added extension 'record_vmenu' priority 4 to asterisk_guitools
  274. -- Added extension 'record_vmenu' priority 3 to asterisk_guitools
  275. -- Added extension 'record_vmenu' priority 2 to asterisk_guitools
  276. -- Added extension 'record_vmenu' priority 1 to asterisk_guitools
  277. -- Added extension 'executecommand' priority 2 to asterisk_guitools
  278. -- Added extension 'executecommand' priority 1 to asterisk_guitools
  279. -- Registered extension context 'pagegroups'; registrar: pbx_config
  280. -- merging incls/swits/igpats from old(pagegroups) to new(pagegroups) context, registrar = pbx_ael
  281. -- Registered extension context 'page_an_extension'; registrar: pbx_config
  282. -- merging incls/swits/igpats from old(page_an_extension) to new(page_an_extension) context, registrar = pbx_ael
  283. -- Registered extension context 'directory'; registrar: pbx_config
  284. -- merging incls/swits/igpats from old(directory) to new(directory) context, registrar = pbx_ael
  285. -- Registered extension context 'voicemailgroups'; registrar: pbx_config
  286. -- merging incls/swits/igpats from old(voicemailgroups) to new(voicemailgroups) context, registrar = pbx_ael
  287. -- Registered extension context 'voicemenus'; registrar: pbx_config
  288. -- merging incls/swits/igpats from old(voicemenus) to new(voicemenus) context, registrar = pbx_ael
  289. -- Registered extension context 'queues'; registrar: pbx_config
  290. -- merging incls/swits/igpats from old(queues) to new(queues) context, registrar = pbx_ael
  291. -- Registered extension context 'ringgroups'; registrar: pbx_config
  292. -- merging incls/swits/igpats from old(ringgroups) to new(ringgroups) context, registrar = pbx_ael
  293. -- Registered extension context 'conferences'; registrar: pbx_config
  294. -- merging incls/swits/igpats from old(conferences) to new(conferences) context, registrar = pbx_ael
  295. -- Registered extension context 'macro-pagingintercom'; registrar: pbx_config
  296. -- merging incls/swits/igpats from old(macro-pagingintercom) to new(macro-pagingintercom) context, registrar = pbx_ael
  297. -- Added extension 's' priority 3 to macro-pagingintercom
  298. -- Added extension 's' priority 2 to macro-pagingintercom
  299. -- Added extension 's' priority 1 to macro-pagingintercom
  300. -- Registered extension context 'macro-stdexten-followme'; registrar: pbx_config
  301. -- merging incls/swits/igpats from old(macro-stdexten-followme) to new(macro-stdexten-followme) context, registrar = pbx_ael
  302. -- Added extension 'a' priority 1 to macro-stdexten-followme
  303. -- Added extension '_s-.' priority 1 to macro-stdexten-followme
  304. -- Added extension 's-BUSY' priority 2 to macro-stdexten-followme
  305. -- Added extension 's-BUSY' priority 1 to macro-stdexten-followme
  306. -- Added extension 's-NOANSWER' priority 1 to macro-stdexten-followme
  307. -- Added extension 's' priority 7 to macro-stdexten-followme
  308. -- Added extension 's' priority 6 to macro-stdexten-followme
  309. -- Added extension 's' priority 5 to macro-stdexten-followme
  310. -- Added extension 's' priority 4 to macro-stdexten-followme
  311. -- Added extension 's' priority 3 to macro-stdexten-followme
  312. -- Added extension 's' priority 2 to macro-stdexten-followme
  313. -- Added extension 's' priority 1 to macro-stdexten-followme
  314. -- Registered extension context 'macro-stdexten'; registrar: pbx_config
  315. -- merging incls/swits/igpats from old(macro-stdexten) to new(macro-stdexten) context, registrar = pbx_ael
  316. -- Added extension 'a' priority 1 to macro-stdexten
  317. -- Added extension '_s-.' priority 1 to macro-stdexten
  318.  
  319. asteros*CLI>
  320.  -- Added extension 's-BUSY' priority 2 to macro-stdexten
  321. -- Added extension 's-BUSY' priority 1 to macro-stdexten
  322.  
  323. asteros*CLI>
  324. 
  325. <--- SIP read from UDP:192.168.127.102:4134 --->
  326. INVITE sip:202@192.168.127.183 SIP/2.0
  327. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-f4078408dd848dfb-1---d8754z-;rport
  328. Max-Forwards: 70
  329. Contact: <sip:201@192.168.127.102:4134>
  330. To: <sip:202@192.168.127.183>
  331. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  332. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  333. CSeq: 1 INVITE
  334. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  335. Content-Type: application/sdp
  336. Supported: replaces
  337. User-Agent: X-Lite release 4.5 stamp 69607
  338. Content-Length: 249
  339.  
  340. v=0
  341. o=- 13012520590148335 1 IN IP4 192.168.127.102
  342. s=X-Lite 4 release 4.5 stamp 69607
  343. c=IN IP4 192.168.127.102
  344. t=0 0
  345. m=audio 49944 RTP/AVP 9 0 8 100 101
  346. a=rtpmap:100 speex/16000
  347. a=rtpmap:101 telephone-event/8000
  348. a=fmtp:101 0-15
  349. a=sendrecv
  350. <------------->
  351. --- (13 headers 10 lines) ---
  352. Sending to 192.168.127.102:4134 (no NAT)
  353. Using INVITE request as basis request - MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  354. Found peer '201' for '201' from 192.168.127.102:4134
  355.  
  356. <--- Reliably Transmitting (no NAT) to 192.168.127.102:4134 --->
  357. SIP/2.0 401 Unauthorized
  358. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-f4078408dd848dfb-1---d8754z-;received=192.168.127.102;rport=4134
  359. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  360. To: <sip:202@192.168.127.183>;tag=as6e7c172b
  361. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  362. CSeq: 1 INVITE
  363. Server: Asterisk PBX 11.3.0
  364. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  365. Supported: replaces, timer
  366. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2329f423"
  367. Content-Length: 0
  368.  
  369.  
  370. <------------>
  371. Scheduling destruction of SIP dialog 'MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY' in 32000 ms (Method: INVITE)
  372.  
  373. asteros*CLI>
  374. 
  375. <--- SIP read from UDP:192.168.127.102:4134 --->
  376. ACK sip:202@192.168.127.183 SIP/2.0
  377. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-f4078408dd848dfb-1---d8754z-;rport
  378. Max-Forwards: 70
  379. To: <sip:202@192.168.127.183>;tag=as6e7c172b
  380. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  381. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  382. CSeq: 1 ACK
  383. Content-Length: 0
  384.  
  385. <------------->
  386. --- (8 headers 0 lines) ---
  387.  
  388. asteros*CLI>
  389. 
  390. <--- SIP read from UDP:192.168.127.102:4134 --->
  391. INVITE sip:202@192.168.127.183 SIP/2.0
  392. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-ac1eaae7effe8ec7-1---d8754z-;rport
  393. Max-Forwards: 70
  394. Contact: <sip:201@192.168.127.102:4134>
  395. To: <sip:202@192.168.127.183>
  396. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  397. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  398. CSeq: 2 INVITE
  399. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  400. Content-Type: application/sdp
  401. Supported: replaces
  402. User-Agent: X-Lite release 4.5 stamp 69607
  403. Authorization: Digest username="201",realm="asterisk",nonce="2329f423",uri="sip:202@192.168.127.183",response="bde607dcc7af399c151e9411bf4f314d",algorithm=MD5
  404. Content-Length: 249
  405.  
  406. v=0
  407. o=- 13012520590148335 1 IN IP4 192.168.127.102
  408. s=X-Lite 4 release 4.5 stamp 69607
  409. c=IN IP4 192.168.127.102
  410. t=0 0
  411. m=audio 49944 RTP/AVP 9 0 8 100 101
  412. a=rtpmap:100 speex/16000
  413. a=rtpmap:101 telephone-event/8000
  414. a=fmtp:101 0-15
  415. a=sendrecv
  416. <------------->
  417.  
  418. asteros*CLI>
  419. --- (14 headers 10 lines) ---
  420.  
  421. asteros*CLI>
  422. Sending to 192.168.127.102:4134 (no NAT)
  423.  
  424. asteros*CLI>
  425. Using INVITE request as basis request - MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  426.  
  427. asteros*CLI>
  428. Found peer '201' for '201' from 192.168.127.102:4134
  429.  
  430. asteros*CLI>
  431.  == Using SIP RTP CoS mark 5
  432.  
  433. asteros*CLI>
  434. Found RTP audio format 9
  435.  
  436. asteros*CLI>
  437. Found RTP audio format 0
  438.  
  439. asteros*CLI>
  440. Found RTP audio format 8
  441.  
  442. asteros*CLI>
  443. Found RTP audio format 100
  444.  
  445. asteros*CLI>
  446. Found RTP audio format 101
  447.  
  448. asteros*CLI>
  449. Found audio description format speex for ID 100
  450.  
  451. asteros*CLI>
  452. Found audio description format telephone-event for ID 101
  453.  
  454. asteros*CLI>
  455. Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|speex16|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  456.  
  457. asteros*CLI>
  458. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  459.  
  460. asteros*CLI>
  461. Peer audio RTP is at port 192.168.127.102:49944
  462.  
  463. asteros*CLI>
  464. Looking for 202 in internal (domain 192.168.127.183)
  465.  
  466. asteros*CLI>
  467. list_route: hop: <sip:201@192.168.127.102:4134>
  468.  
  469. asteros*CLI>
  470. 
  471. <--- Transmitting (no NAT) to 192.168.127.102:4134 --->
  472. SIP/2.0 100 Trying
  473. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-ac1eaae7effe8ec7-1---d8754z-;received=192.168.127.102;rport=4134
  474. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  475. To: <sip:202@192.168.127.183>
  476. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  477. CSeq: 2 INVITE
  478. Server: Asterisk PBX 11.3.0
  479. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  480. Supported: replaces, timer
  481. Contact: <sip:202@192.168.127.183:5060>
  482. Content-Length: 0
  483.  
  484.  
  485. <------------>
  486.  
  487. asteros*CLI>
  488.  -- Executing [202@internal:1] Dial("SIP/201-00000060", "SIP/202,15,tT") in new stack
  489.  
  490. asteros*CLI>
  491.  == Using SIP RTP CoS mark 5
  492.  
  493. asteros*CLI>
  494. Audio is at 10242
  495.  
  496. asteros*CLI>
  497. Adding codec 100003 (ulaw) to SDP
  498.  
  499. asteros*CLI>
  500. Adding codec 100002 (gsm) to SDP
  501.  
  502. asteros*CLI>
  503. Adding codec 100004 (alaw) to SDP
  504.  
  505. asteros*CLI>
  506. Adding codec 100017 (testlaw) to SDP
  507.  
  508. asteros*CLI>
  509. Adding non-codec 0x1 (telephone-event) to SDP
  510.  
  511. asteros*CLI>
  512. Reliably Transmitting (no NAT) to 192.168.127.138:64668:
  513. INVITE sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752 SIP/2.0
  514. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK34691c1e
  515. Max-Forwards: 70
  516. From: "Brian Salazar" <sip:201@192.168.127.183>;tag=as3257e50c
  517. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  518. Contact: <sip:201@192.168.127.183:5060>
  519. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  520. CSeq: 102 INVITE
  521. User-Agent: Asterisk PBX 11.3.0
  522. Date: Tue, 07 May 2013 18:04:29 GMT
  523. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  524. Supported: replaces, timer
  525. Content-Type: application/sdp
  526. Content-Length: 315
  527.  
  528. v=0
  529. o=root 1272138234 1272138234 IN IP4 192.168.127.183
  530. s=Asterisk PBX 11.3.0
  531. c=IN IP4 192.168.127.183
  532. t=0 0
  533. m=audio 10242 RTP/AVP 0 3 8 101
  534. a=rtpmap:0 PCMU/8000
  535. a=rtpmap:3 GSM/8000
  536. a=rtpmap:8 PCMA/8000
  537. a=rtpmap:101 telephone-event/8000
  538. a=fmtp:101 0-16
  539. a=silenceSupp:off - - - -
  540. a=ptime:20
  541. a=sendrecv
  542.  
  543. ---
  544.  
  545. asteros*CLI>
  546.  -- Called SIP/202
  547.  
  548. asteros*CLI>
  549. Retransmitting #1 (no NAT) to 192.168.127.138:64668:
  550. INVITE sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752 SIP/2.0
  551. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK34691c1e
  552. Max-Forwards: 70
  553. From: "Brian Salazar" <sip:201@192.168.127.183>;tag=as3257e50c
  554. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  555. Contact: <sip:201@192.168.127.183:5060>
  556. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  557. CSeq: 102 INVITE
  558. User-Agent: Asterisk PBX 11.3.0
  559. Date: Tue, 07 May 2013 18:04:29 GMT
  560. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  561. Supported: replaces, timer
  562. Content-Type: application/sdp
  563. Content-Length: 315
  564.  
  565. v=0
  566. o=root 1272138234 1272138234 IN IP4 192.168.127.183
  567. s=Asterisk PBX 11.3.0
  568. c=IN IP4 192.168.127.183
  569. t=0 0
  570. m=audio 10242 RTP/AVP 0 3 8 101
  571. a=rtpmap:0 PCMU/8000
  572. a=rtpmap:3 GSM/8000
  573. a=rtpmap:8 PCMA/8000
  574. a=rtpmap:101 telephone-event/8000
  575. a=fmtp:101 0-16
  576. a=silenceSupp:off - - - -
  577. a=ptime:20
  578. a=sendrecv
  579.  
  580. ---
  581.  
  582. asteros*CLI>
  583. 
  584. <--- SIP read from UDP:192.168.127.138:64668 --->
  585. SIP/2.0 180 Ringing
  586. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK34691c1e
  587. Contact: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  588. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  589. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=as3257e50c
  590. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  591. CSeq: 102 INVITE
  592. User-Agent: X-Lite release 1011s stamp 41150
  593. Content-Length: 0
  594.  
  595. <------------->
  596. --- (9 headers 0 lines) ---
  597. list_route: hop: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  598.  
  599. <--- SIP read from UDP:192.168.127.138:64668 --->
  600. SIP/2.0 180 Ringing
  601. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK34691c1e
  602. Contact: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  603. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  604. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=as3257e50c
  605. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  606. CSeq: 102 INVITE
  607. User-Agent: X-Lite release 1011s stamp 41150
  608. Content-Length: 0
  609.  
  610. <------------->
  611. --- (9 headers 0 lines) ---
  612. list_route: hop: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  613.  
  614. asteros*CLI>
  615. 
  616. <--- SIP read from UDP:192.168.127.138:64668 --->
  617.  
  618.  
  619. <------------->
  620.  
  621. asteros*CLI>
  622.  -- SIP/202-00000061 is ringing
  623.  
  624. asteros*CLI>
  625. 
  626. <--- Transmitting (no NAT) to 192.168.127.102:4134 --->
  627. SIP/2.0 180 Ringing
  628. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-ac1eaae7effe8ec7-1---d8754z-;received=192.168.127.102;rport=4134
  629. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  630. To: <sip:202@192.168.127.183>;tag=as2b0ff98d
  631. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  632. CSeq: 2 INVITE
  633. Server: Asterisk PBX 11.3.0
  634. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  635. Supported: replaces, timer
  636. Contact: <sip:202@192.168.127.183:5060>
  637. Content-Length: 0
  638.  
  639.  
  640. <------------>
  641.  
  642. asteros*CLI>
  643.  -- SIP/202-00000061 is ringing
  644.  
  645. asteros*CLI>
  646. 
  647. <--- SIP read from UDP:192.168.127.108:7572 --->
  648.  
  649.  
  650. <------------->
  651.  
  652. asteros*CLI>
  653. 
  654. <--- SIP read from UDP:192.168.127.138:64668 --->
  655. SIP/2.0 200 OK
  656. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK34691c1e
  657. Contact: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  658. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  659. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=as3257e50c
  660. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  661. CSeq: 102 INVITE
  662. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  663. Content-Type: application/sdp
  664. User-Agent: X-Lite release 1011s stamp 41150
  665. Content-Length: 191
  666.  
  667. v=0
  668. o=- 6 2 IN IP4 192.168.127.138
  669. s=CounterPath X-Lite 3.0
  670. c=IN IP4 192.168.127.138
  671. t=0 0
  672. m=audio 32170 RTP/AVP 0 8 101
  673. a=fmtp:101 0-15
  674. a=rtpmap:101 telephone-event/8000
  675. a=sendrecv
  676. <------------->
  677. --- (11 headers 9 lines) ---
  678. Found RTP audio format 0
  679. Found RTP audio format 8
  680. Found RTP audio format 101
  681. Found audio description format telephone-event for ID 101
  682. Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  683. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  684. Peer audio RTP is at port 192.168.127.138:32170
  685. list_route: hop: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  686. set_destination: Parsing <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752> for address/port to send to
  687. set_destination: set destination to 192.168.127.138:64668
  688. Transmitting (no NAT) to 192.168.127.138:64668:
  689. ACK sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752 SIP/2.0
  690. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK33f5dac7
  691. Max-Forwards: 70
  692. From: "Brian Salazar" <sip:201@192.168.127.183>;tag=as3257e50c
  693. To: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  694. Contact: <sip:201@192.168.127.183:5060>
  695. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  696. CSeq: 102 ACK
  697. User-Agent: Asterisk PBX 11.3.0
  698. Content-Length: 0
  699.  
  700.  
  701. ---
  702.  
  703. asteros*CLI>
  704.  -- SIP/202-00000061 answered SIP/201-00000060
  705. Audio is at 15082
  706. Adding codec 100003 (ulaw) to SDP
  707. Adding codec 100004 (alaw) to SDP
  708. Adding non-codec 0x1 (telephone-event) to SDP
  709.  
  710. <--- Reliably Transmitting (no NAT) to 192.168.127.102:4134 --->
  711. SIP/2.0 200 OK
  712. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-ac1eaae7effe8ec7-1---d8754z-;received=192.168.127.102;rport=4134
  713. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  714. To: <sip:202@192.168.127.183>;tag=as2b0ff98d
  715. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  716. CSeq: 2 INVITE
  717. Server: Asterisk PBX 11.3.0
  718. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  719. Supported: replaces, timer
  720. Contact: <sip:202@192.168.127.183:5060>
  721. Content-Type: application/sdp
  722. Content-Length: 290
  723.  
  724. v=0
  725. o=root 272217756 272217756 IN IP4 192.168.127.183
  726. s=Asterisk PBX 11.3.0
  727. c=IN IP4 192.168.127.183
  728. t=0 0
  729. m=audio 15082 RTP/AVP 0 8 101
  730. a=rtpmap:0 PCMU/8000
  731. a=rtpmap:8 PCMA/8000
  732. a=rtpmap:101 telephone-event/8000
  733. a=fmtp:101 0-16
  734. a=silenceSupp:off - - - -
  735. a=ptime:20
  736. a=sendrecv
  737.  
  738. <------------>
  739.  
  740. asteros*CLI>
  741. 
  742. <--- SIP read from UDP:192.168.127.102:4134 --->
  743. ACK sip:202@192.168.127.183:5060 SIP/2.0
  744. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-db9bef8c38a368f6-1---d8754z-;rport
  745. Max-Forwards: 70
  746. Contact: <sip:201@192.168.127.102:4134>
  747. To: <sip:202@192.168.127.183>;tag=as2b0ff98d
  748. From: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  749. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  750. CSeq: 2 ACK
  751. User-Agent: X-Lite release 4.5 stamp 69607
  752. Content-Length: 0
  753.  
  754. <------------->
  755.  
  756. asteros*CLI>
  757. --- (10 headers 0 lines) ---
  758.  
  759. asteros*CLI>
  760. 
  761. <--- SIP read from UDP:192.168.127.138:64668 --->
  762. BYE sip:201@192.168.127.183:5060 SIP/2.0
  763. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-95721d1dc52fa301-1--d87543-;rport
  764. Max-Forwards: 70
  765. Contact: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>
  766. To: "Brian Salazar"<sip:201@192.168.127.183>;tag=as3257e50c
  767. From: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  768. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  769. CSeq: 2 BYE
  770. User-Agent: X-Lite release 1011s stamp 41150
  771. Reason: SIP;description="User Hung Up"
  772. Content-Length: 0
  773.  
  774. <------------->
  775. --- (11 headers 0 lines) ---
  776. Sending to 192.168.127.138:64668 (no NAT)
  777. Scheduling destruction of SIP dialog '323caa12109de0292075762b14c38ec9@192.168.127.183:5060' in 32000 ms (Method: BYE)
  778.  
  779. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  780. SIP/2.0 200 OK
  781. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-95721d1dc52fa301-1--d87543-;received=192.168.127.138;rport=64668
  782. From: <sip:202@192.168.127.138:64668;rinstance=18a66403bdfc3752>;tag=71010e24
  783. To: "Brian Salazar"<sip:201@192.168.127.183>;tag=as3257e50c
  784. Call-ID: 323caa12109de0292075762b14c38ec9@192.168.127.183:5060
  785. CSeq: 2 BYE
  786. Server: Asterisk PBX 11.3.0
  787. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  788. Supported: replaces, timer
  789. Content-Length: 0
  790.  
  791.  
  792. <------------>
  793.  
  794. asteros*CLI>
  795.  == Spawn extension (internal, 202, 1) exited non-zero on 'SIP/201-00000060'
  796.  
  797. asteros*CLI>
  798. Scheduling destruction of SIP dialog 'MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY' in 32000 ms (Method: ACK)
  799.  
  800. asteros*CLI>
  801. set_destination: Parsing <sip:201@192.168.127.102:4134> for address/port to send to
  802.  
  803. asteros*CLI>
  804. set_destination: set destination to 192.168.127.102:4134
  805.  
  806. asteros*CLI>
  807. Reliably Transmitting (no NAT) to 192.168.127.102:4134:
  808. BYE sip:201@192.168.127.102:4134 SIP/2.0
  809. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK184533f3;rport
  810. Max-Forwards: 70
  811. From: <sip:202@192.168.127.183>;tag=as2b0ff98d
  812. To: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  813. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  814. CSeq: 102 BYE
  815. User-Agent: Asterisk PBX 11.3.0
  816. Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:192.168.127.183", nonce="2329f423", response="b1af61b3fa28acd09852a79e7660ca4c"
  817. X-Asterisk-HangupCause: Normal Clearing
  818. X-Asterisk-HangupCauseCode: 16
  819. Content-Length: 0
  820.  
  821.  
  822. ---
  823.  
  824. asteros*CLI>
  825. 
  826. <--- SIP read from UDP:192.168.127.102:4134 --->
  827. SIP/2.0 200 OK
  828. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK184533f3;rport=5060
  829. Contact: <sip:201@192.168.127.102:4134>
  830. To: "Brian Salazar"<sip:201@192.168.127.183>;tag=2558bd45
  831. From: <sip:202@192.168.127.183>;tag=as2b0ff98d
  832. Call-ID: MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY
  833. CSeq: 102 BYE
  834. User-Agent: X-Lite release 4.5 stamp 69607
  835. Content-Length: 0
  836.  
  837. <------------->
  838. --- (9 headers 0 lines) ---
  839. SIP Response message for INCOMING dialog BYE arrived
  840. Really destroying SIP dialog 'MzBjYzMyNGMxNDExZjc2NTBkYzQxNjAwMWVmZTJmYjY' Method: ACK
  841.  
  842. asteros*CLI>
  843. 
  844. <--- SIP read from UDP:192.168.127.102:4134 --->
  845.  
  846.  
  847. <------------->
  848.  
  849. asteros*CLI>
  850. 
  851. <--- SIP read from UDP:192.168.127.138:64668 --->
  852. SUBSCRIBE sip:asterisk@192.168.127.183:5060 SIP/2.0
  853. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-fa29757505789f18-1--d87543-;rport
  854. Max-Forwards: 70
  855. Contact: <sip:202@192.168.127.138:64668>
  856. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=as1cb8f684
  857. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=5b725930
  858. Call-ID: NjIxYjE0ZTE3ZGY1NTIxYWVlNzVjZjNhYTZlZTE4MTQ.
  859. CSeq: 4 SUBSCRIBE
  860. Expires: 300
  861. User-Agent: X-Lite release 1011s stamp 41150
  862. Authorization: Digest username="202",realm="asterisk",nonce="78f28e36",uri="sip:asterisk@192.168.127.183:5060",response="e366169ce35405cb24eacdbd5b3423a6",algorithm=MD5
  863. Event: message-summary
  864. Content-Length: 0
  865.  
  866. <------------->
  867. --- (13 headers 0 lines) ---
  868.  
  869. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  870. SIP/2.0 481 Call/Transaction Does Not Exist
  871. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-fa29757505789f18-1--d87543-;rport;received=192.168.127.138
  872. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=5b725930
  873. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=as1cb8f684
  874. Call-ID: NjIxYjE0ZTE3ZGY1NTIxYWVlNzVjZjNhYTZlZTE4MTQ.
  875. CSeq: 4 SUBSCRIBE
  876. Server: Asterisk PBX 11.3.0
  877. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  878. Supported: replaces, timer
  879. Content-Length: 0
  880.  
  881.  
  882. <------------>
  883. Scheduling destruction of SIP dialog 'NjIxYjE0ZTE3ZGY1NTIxYWVlNzVjZjNhYTZlZTE4MTQ.' in 32000 ms (Method: SUBSCRIBE)
  884.  
  885. asteros*CLI>
  886. 
  887. <--- SIP read from UDP:192.168.127.138:64668 --->
  888. SUBSCRIBE sip:202@192.168.127.183 SIP/2.0
  889. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-b361ac50d0581015-1--d87543-;rport
  890. Max-Forwards: 70
  891. Contact: <sip:202@192.168.127.138:64668>
  892. To: "Brian Salazar"<sip:202@192.168.127.183>
  893. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  894. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  895. CSeq: 1 SUBSCRIBE
  896. Expires: 300
  897. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  898. User-Agent: X-Lite release 1011s stamp 41150
  899. Event: message-summary
  900. Content-Length: 0
  901.  
  902. <------------->
  903. --- (13 headers 0 lines) ---
  904. Creating new subscription
  905. Sending to 192.168.127.138:64668 (no NAT)
  906. list_route: hop: <sip:202@192.168.127.138:64668>
  907. Found peer '202' for '202' from 192.168.127.138:64668
  908.  
  909. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  910. SIP/2.0 401 Unauthorized
  911. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-b361ac50d0581015-1--d87543-;received=192.168.127.138;rport=64668
  912. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  913. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=as6da66b13
  914. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  915. CSeq: 1 SUBSCRIBE
  916. Server: Asterisk PBX 11.3.0
  917. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  918. Supported: replaces, timer
  919. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d8825be"
  920. Content-Length: 0
  921.  
  922.  
  923. <------------>
  924. Scheduling destruction of SIP dialog 'MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.' in 32000 ms (Method: SUBSCRIBE)
  925.  
  926. <--- SIP read from UDP:192.168.127.138:64668 --->
  927.  
  928.  
  929. <------------->
  930.  
  931. asteros*CLI>
  932. 
  933. <--- SIP read from UDP:192.168.127.138:64668 --->
  934. SUBSCRIBE sip:202@192.168.127.183 SIP/2.0
  935. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-b361ac50d0581015-1--d87543-;rport
  936. Max-Forwards: 70
  937. Contact: <sip:202@192.168.127.138:64668>
  938. To: "Brian Salazar"<sip:202@192.168.127.183>
  939. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  940. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  941. CSeq: 1 SUBSCRIBE
  942. Expires: 300
  943. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  944. User-Agent: X-Lite release 1011s stamp 41150
  945. Event: message-summary
  946. Content-Length: 0
  947.  
  948. <------------->
  949.  
  950. asteros*CLI>
  951. --- (13 headers 0 lines) ---
  952.  
  953. asteros*CLI>
  954. Ignoring this SUBSCRIBE request
  955.  
  956. asteros*CLI>
  957. Found peer '202' for '202' from 192.168.127.138:64668
  958.  
  959. asteros*CLI>
  960. 
  961. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  962. SIP/2.0 401 Unauthorized
  963. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-b361ac50d0581015-1--d87543-;received=192.168.127.138;rport=64668
  964. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  965. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=as6da66b13
  966. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  967. CSeq: 1 SUBSCRIBE
  968. Server: Asterisk PBX 11.3.0
  969. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  970. Supported: replaces, timer
  971. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d8825be"
  972. Content-Length: 0
  973.  
  974.  
  975. <------------>
  976.  
  977. asteros*CLI>
  978. Scheduling destruction of SIP dialog 'MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.' in 32000 ms (Method: SUBSCRIBE)
  979.  
  980. asteros*CLI>
  981. 
  982. <--- SIP read from UDP:192.168.127.108:7572 --->
  983.  
  984.  
  985. <------------->
  986.  
  987. asteros*CLI>
  988. 
  989. <--- SIP read from UDP:192.168.127.138:64668 --->
  990. SUBSCRIBE sip:202@192.168.127.183 SIP/2.0
  991. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-1567e878125ff567-1--d87543-;rport
  992. Max-Forwards: 70
  993. Contact: <sip:202@192.168.127.138:64668>
  994. To: "Brian Salazar"<sip:202@192.168.127.183>
  995. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  996. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  997. CSeq: 2 SUBSCRIBE
  998. Expires: 300
  999. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  1000. User-Agent: X-Lite release 1011s stamp 41150
  1001. Authorization: Digest username="202",realm="asterisk",nonce="6d8825be",uri="sip:202@192.168.127.183",response="2ba5c50e7dd471fe50347d3af6aafaeb",algorithm=MD5
  1002. Event: message-summary
  1003. Content-Length: 0
  1004.  
  1005. <------------->
  1006. --- (14 headers 0 lines) ---
  1007. Creating new subscription
  1008. Sending to 192.168.127.138:64668 (no NAT)
  1009. Found peer '202' for '202' from 192.168.127.138:64668
  1010. Scheduling destruction of SIP dialog 'MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.' in 310000 ms (Method: SUBSCRIBE)
  1011.  
  1012. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  1013. SIP/2.0 200 OK
  1014. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-1567e878125ff567-1--d87543-;received=192.168.127.138;rport=64668
  1015. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  1016. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=as6da66b13
  1017. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1018. CSeq: 2 SUBSCRIBE
  1019. Server: Asterisk PBX 11.3.0
  1020. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1021. Supported: replaces, timer
  1022. Expires: 300
  1023. Contact: <sip:202@192.168.127.183:5060>;expires=300
  1024. Content-Length: 0
  1025.  
  1026.  
  1027. <------------>
  1028. Reliably Transmitting (no NAT) to 192.168.127.138:64668:
  1029. NOTIFY sip:202@192.168.127.138:64668 SIP/2.0
  1030. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK431be95e
  1031. Max-Forwards: 70
  1032. Route: <sip:202@192.168.127.138:64668>
  1033. From: "asterisk" <sip:asterisk@192.168.127.183>;tag=as6da66b13
  1034. To: <sip:202@192.168.127.138:64668>;tag=b759475f
  1035. Contact: <sip:asterisk@192.168.127.183:5060>
  1036. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1037. CSeq: 102 NOTIFY
  1038. User-Agent: Asterisk PBX 11.3.0
  1039. Event: message-summary
  1040. Content-Type: application/simple-message-summary
  1041. Subscription-State: active
  1042. Content-Length: 96
  1043.  
  1044. Messages-Waiting: yes
  1045. Message-Account: sip:asterisk@192.168.127.183
  1046. Voice-Message: 2/2 (0/0)
  1047.  
  1048. ---
  1049.  
  1050. asteros*CLI>
  1051. 
  1052. <--- SIP read from UDP:192.168.127.138:64668 --->
  1053. SUBSCRIBE sip:202@192.168.127.183 SIP/2.0
  1054. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-1567e878125ff567-1--d87543-;rport
  1055. Max-Forwards: 70
  1056. Contact: <sip:202@192.168.127.138:64668>
  1057. To: "Brian Salazar"<sip:202@192.168.127.183>
  1058. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  1059. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1060. CSeq: 2 SUBSCRIBE
  1061. Expires: 300
  1062. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  1063. User-Agent: X-Lite release 1011s stamp 41150
  1064. Authorization: Digest username="202",realm="asterisk",nonce="6d8825be",uri="sip:202@192.168.127.183",response="2ba5c50e7dd471fe50347d3af6aafaeb",algorithm=MD5
  1065. Event: message-summary
  1066. Content-Length: 0
  1067.  
  1068. <------------->
  1069.  
  1070. asteros*CLI>
  1071. --- (14 headers 0 lines) ---
  1072.  
  1073. asteros*CLI>
  1074. Ignoring this SUBSCRIBE request
  1075.  
  1076. asteros*CLI>
  1077. 
  1078. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  1079. SIP/2.0 404 Not found
  1080. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-1567e878125ff567-1--d87543-;received=192.168.127.138;rport=64668
  1081. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  1082. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=b759475f
  1083. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1084. CSeq: 2 SUBSCRIBE
  1085. Server: Asterisk PBX 11.3.0
  1086. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1087. Supported: replaces, timer
  1088. Content-Length: 0
  1089.  
  1090.  
  1091. <------------>
  1092.  
  1093. asteros*CLI>
  1094. Retransmitting #1 (no NAT) to 192.168.127.138:64668:
  1095. NOTIFY sip:202@192.168.127.138:64668 SIP/2.0
  1096. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK431be95e
  1097. Max-Forwards: 70
  1098. Route: <sip:202@192.168.127.138:64668>
  1099. From: "asterisk" <sip:asterisk@192.168.127.183>;tag=as6da66b13
  1100. To: <sip:202@192.168.127.138:64668>;tag=b759475f
  1101. Contact: <sip:asterisk@192.168.127.183:5060>
  1102. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1103. CSeq: 102 NOTIFY
  1104. User-Agent: Asterisk PBX 11.3.0
  1105. Event: message-summary
  1106. Content-Type: application/simple-message-summary
  1107. Subscription-State: active
  1108. Content-Length: 96
  1109.  
  1110. Messages-Waiting: yes
  1111. Message-Account: sip:asterisk@192.168.127.183
  1112. Voice-Message: 2/2 (0/0)
  1113.  
  1114. ---
  1115.  
  1116. asteros*CLI>
  1117. 
  1118. <--- SIP read from UDP:192.168.127.138:64668 --->
  1119. SIP/2.0 200 OK
  1120. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK431be95e
  1121. Contact: <sip:202@192.168.127.138:64668>
  1122. To: <sip:202@192.168.127.138:64668>;tag=b759475f
  1123. From: "asterisk"<sip:asterisk@192.168.127.183>;tag=as6da66b13
  1124. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1125. CSeq: 102 NOTIFY
  1126. User-Agent: X-Lite release 1011s stamp 41150
  1127. Content-Length: 0
  1128.  
  1129. <------------->
  1130. --- (9 headers 0 lines) ---
  1131. Really destroying SIP dialog 'MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.' Method: SUBSCRIBE
  1132.  
  1133. <--- SIP read from UDP:192.168.127.138:64668 --->
  1134. SIP/2.0 200 OK
  1135. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK431be95e
  1136. Contact: <sip:202@192.168.127.138:64668>
  1137. To: <sip:202@192.168.127.138:64668>;tag=b759475f
  1138. From: "asterisk"<sip:asterisk@192.168.127.183>;tag=as6da66b13
  1139. Call-ID: MmM2Nzc0OTgxODAyZjZmM2VjOGUyNWQ4ZDhlYjg3MmY.
  1140. CSeq: 102 NOTIFY
  1141. User-Agent: X-Lite release 1011s stamp 41150
  1142. Content-Length: 0
  1143.  
  1144. <------------->
  1145. --- (9 headers 0 lines) ---
  1146.  
  1147. asteros*CLI>
  1148. 
  1149. <--- SIP read from UDP:192.168.127.138:64668 --->
  1150. INVITE sip:201@192.168.127.183 SIP/2.0
  1151. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-da663a4ea1648244-1--d87543-;rport
  1152. Max-Forwards: 70
  1153. Contact: <sip:202@192.168.127.138:64668>
  1154. To: "201"<sip:201@192.168.127.183>
  1155. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1156. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1157. CSeq: 1 INVITE
  1158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  1159. Content-Type: application/sdp
  1160. User-Agent: X-Lite release 1011s stamp 41150
  1161. Content-Length: 531
  1162.  
  1163. v=0
  1164. o=- 1 2 IN IP4 192.168.127.138
  1165. s=CounterPath X-Lite 3.0
  1166. c=IN IP4 192.168.127.138
  1167. t=0 0
  1168. m=audio 25136 RTP/AVP 107 119 100 106 0 105 98 8 101
  1169. a=alt:1 3 : Thptiy2g vHYmFs6h 192.168.160.1 25136
  1170. a=alt:2 2 : IFp8uNWC vzy9usQt 192.168.174.1 25136
  1171. a=alt:3 1 : F/f0Ejcm YDPVBWf3 192.168.127.138 25136
  1172. a=fmtp:101 0-15
  1173. a=rtpmap:107 BV32/16000
  1174. a=rtpmap:119 BV32-FEC/16000
  1175. a=rtpmap:100 SPEEX/16000
  1176. a=rtpmap:106 SPEEX-FEC/16000
  1177. a=rtpmap:105 SPEEX-FEC/8000
  1178. a=rtpmap:98 iLBC/8000
  1179. a=rtpmap:101 telephone-event/8000
  1180. a=sendrecv
  1181. <------------->
  1182. --- (12 headers 18 lines) ---
  1183. Sending to 192.168.127.138:64668 (no NAT)
  1184. Using INVITE request as basis request - NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1185. Found peer '202' for '202' from 192.168.127.138:64668
  1186.  
  1187. <--- Reliably Transmitting (no NAT) to 192.168.127.138:64668 --->
  1188. SIP/2.0 401 Unauthorized
  1189. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-da663a4ea1648244-1--d87543-;received=192.168.127.138;rport=64668
  1190. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1191. To: "201"<sip:201@192.168.127.183>;tag=as07220dc3
  1192. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1193. CSeq: 1 INVITE
  1194. Server: Asterisk PBX 11.3.0
  1195. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1196. Supported: replaces, timer
  1197. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c084d9d"
  1198. Content-Length: 0
  1199.  
  1200.  
  1201. <------------>
  1202. Scheduling destruction of SIP dialog 'NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.' in 32000 ms (Method: INVITE)
  1203.  
  1204. asteros*CLI>
  1205. 
  1206. <--- SIP read from UDP:192.168.127.138:64668 --->
  1207. ACK sip:201@192.168.127.183 SIP/2.0
  1208. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-da663a4ea1648244-1--d87543-;rport
  1209. To: "201"<sip:201@192.168.127.183>;tag=as07220dc3
  1210. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1211. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1212. CSeq: 1 ACK
  1213. Content-Length: 0
  1214.  
  1215. <------------->
  1216. --- (7 headers 0 lines) ---
  1217.  
  1218. asteros*CLI>
  1219. 
  1220. <--- SIP read from UDP:192.168.127.138:64668 --->
  1221. INVITE sip:201@192.168.127.183 SIP/2.0
  1222. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-9e34ec0c7a7b965c-1--d87543-;rport
  1223. Max-Forwards: 70
  1224. Contact: <sip:202@192.168.127.138:64668>
  1225. To: "201"<sip:201@192.168.127.183>
  1226. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1227. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1228. CSeq: 2 INVITE
  1229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  1230. Content-Type: application/sdp
  1231. User-Agent: X-Lite release 1011s stamp 41150
  1232. Authorization: Digest username="202",realm="asterisk",nonce="7c084d9d",uri="sip:201@192.168.127.183",response="671262764f3f01fc02063a1510075361",algorithm=MD5
  1233. Content-Length: 531
  1234.  
  1235. v=0
  1236. o=- 1 2 IN IP4 192.168.127.138
  1237. s=CounterPath X-Lite 3.0
  1238. c=IN IP4 192.168.127.138
  1239. t=0 0
  1240. m=audio 25136 RTP/AVP 107 119 100 106 0 105 98 8 101
  1241. a=alt:1 3 : Thptiy2g vHYmFs6h 192.168.160.1 25136
  1242. a=alt:2 2 : IFp8uNWC vzy9usQt 192.168.174.1 25136
  1243. a=alt:3 1 : F/f0Ejcm YDPVBWf3 192.168.127.138 25136
  1244. a=fmtp:101 0-15
  1245. a=rtpmap:107 BV32/16000
  1246. a=rtpmap:119 BV32-FEC/16000
  1247. a=rtpmap:100 SPEEX/16000
  1248. a=rtpmap:106 SPEEX-FEC/16000
  1249. a=rtpmap:105 SPEEX-FEC/8000
  1250. a=rtpmap:98 iLBC/8000
  1251. a=rtpmap:101 telephone-event/8000
  1252. a=sendrecv
  1253. <------------->
  1254.  
  1255. asteros*CLI>
  1256. --- (13 headers 18 lines) ---
  1257.  
  1258. asteros*CLI>
  1259. Sending to 192.168.127.138:64668 (no NAT)
  1260.  
  1261. asteros*CLI>
  1262. Using INVITE request as basis request - NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1263.  
  1264. asteros*CLI>
  1265. Found peer '202' for '202' from 192.168.127.138:64668
  1266.  
  1267. asteros*CLI>
  1268.  == Using SIP RTP CoS mark 5
  1269.  
  1270. asteros*CLI>
  1271. Found RTP audio format 107
  1272.  
  1273. asteros*CLI>
  1274. Found RTP audio format 119
  1275.  
  1276. asteros*CLI>
  1277. Found RTP audio format 100
  1278.  
  1279. asteros*CLI>
  1280. Found RTP audio format 106
  1281.  
  1282. asteros*CLI>
  1283. Found RTP audio format 0
  1284.  
  1285. asteros*CLI>
  1286. Found RTP audio format 105
  1287.  
  1288. asteros*CLI>
  1289. Found RTP audio format 98
  1290.  
  1291. asteros*CLI>
  1292. Found RTP audio format 8
  1293.  
  1294. asteros*CLI>
  1295. Found RTP audio format 101
  1296.  
  1297. asteros*CLI>
  1298. Found unknown media description format BV32 for ID 107
  1299.  
  1300. asteros*CLI>
  1301. Found unknown media description format BV32-FEC for ID 119
  1302.  
  1303. asteros*CLI>
  1304. Found audio description format SPEEX for ID 100
  1305.  
  1306. asteros*CLI>
  1307. Found unknown media description format SPEEX-FEC for ID 106
  1308.  
  1309. asteros*CLI>
  1310. Found unknown media description format SPEEX-FEC for ID 105
  1311.  
  1312. asteros*CLI>
  1313. Found audio description format iLBC for ID 98
  1314.  
  1315. asteros*CLI>
  1316. Found audio description format telephone-event for ID 101
  1317.  
  1318. asteros*CLI>
  1319. Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|speex16|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  1320.  
  1321. asteros*CLI>
  1322. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  1323.  
  1324. asteros*CLI>
  1325. Peer audio RTP is at port 192.168.127.138:25136
  1326.  
  1327. asteros*CLI>
  1328. Looking for 201 in internal (domain 192.168.127.183)
  1329.  
  1330. asteros*CLI>
  1331. list_route: hop: <sip:202@192.168.127.138:64668>
  1332.  
  1333. asteros*CLI>
  1334. 
  1335. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  1336. SIP/2.0 100 Trying
  1337. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-9e34ec0c7a7b965c-1--d87543-;received=192.168.127.138;rport=64668
  1338. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1339. To: "201"<sip:201@192.168.127.183>
  1340. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1341. CSeq: 2 INVITE
  1342. Server: Asterisk PBX 11.3.0
  1343. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1344. Supported: replaces, timer
  1345. Contact: <sip:201@192.168.127.183:5060>
  1346. Content-Length: 0
  1347.  
  1348.  
  1349. <------------>
  1350.  
  1351. asteros*CLI>
  1352.  -- Executing [201@internal:1] Dial("SIP/202-00000062", "SIP/201,15,tT") in new stack
  1353.  
  1354. asteros*CLI>
  1355.  == Using SIP RTP CoS mark 5
  1356.  
  1357. asteros*CLI>
  1358. Audio is at 12260
  1359.  
  1360. asteros*CLI>
  1361. Adding codec 100003 (ulaw) to SDP
  1362.  
  1363. asteros*CLI>
  1364. Adding codec 100002 (gsm) to SDP
  1365.  
  1366. asteros*CLI>
  1367. Adding codec 100004 (alaw) to SDP
  1368.  
  1369. asteros*CLI>
  1370. Adding codec 100017 (testlaw) to SDP
  1371.  
  1372. asteros*CLI>
  1373. Adding non-codec 0x1 (telephone-event) to SDP
  1374.  
  1375. asteros*CLI>
  1376. Reliably Transmitting (no NAT) to 192.168.127.102:4134:
  1377. INVITE sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671 SIP/2.0
  1378. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK3be07969
  1379. Max-Forwards: 70
  1380. From: "Eduardo Salazar" <sip:202@192.168.127.183>;tag=as78bcc2d1
  1381. To: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>
  1382. Contact: <sip:202@192.168.127.183:5060>
  1383. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1384. CSeq: 102 INVITE
  1385. User-Agent: Asterisk PBX 11.3.0
  1386. Date: Tue, 07 May 2013 18:05:04 GMT
  1387. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1388. Supported: replaces, timer
  1389. Content-Type: application/sdp
  1390. Content-Length: 313
  1391.  
  1392. v=0
  1393. o=root 316822457 316822457 IN IP4 192.168.127.183
  1394. s=Asterisk PBX 11.3.0
  1395. c=IN IP4 192.168.127.183
  1396. t=0 0
  1397. m=audio 12260 RTP/AVP 0 3 8 101
  1398. a=rtpmap:0 PCMU/8000
  1399. a=rtpmap:3 GSM/8000
  1400. a=rtpmap:8 PCMA/8000
  1401. a=rtpmap:101 telephone-event/8000
  1402. a=fmtp:101 0-16
  1403. a=silenceSupp:off - - - -
  1404. a=ptime:20
  1405. a=sendrecv
  1406.  
  1407. ---
  1408.  
  1409. asteros*CLI>
  1410.  -- Called SIP/201
  1411.  
  1412. asteros*CLI>
  1413. 
  1414. <--- SIP read from UDP:192.168.127.102:4134 --->
  1415. SIP/2.0 100 Trying
  1416. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK3be07969
  1417. To: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>
  1418. From: "Eduardo Salazar" <sip:202@192.168.127.183>;tag=as78bcc2d1
  1419. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1420. CSeq: 102 INVITE
  1421. Content-Length: 0
  1422.  
  1423. <------------->
  1424. --- (7 headers 0 lines) ---
  1425.  
  1426. asteros*CLI>
  1427. 
  1428. <--- SIP read from UDP:192.168.127.102:4134 --->
  1429. SIP/2.0 180 Ringing
  1430. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK3be07969
  1431. Contact: <sip:201@192.168.127.102:4134>
  1432. To: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>;tag=ececafca
  1433. From: "Eduardo Salazar"<sip:202@192.168.127.183>;tag=as78bcc2d1
  1434. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1435. CSeq: 102 INVITE
  1436. User-Agent: X-Lite release 4.5 stamp 69607
  1437. Content-Length: 0
  1438.  
  1439. <------------->
  1440. --- (9 headers 0 lines) ---
  1441. list_route: hop: <sip:201@192.168.127.102:4134>
  1442.  
  1443. asteros*CLI>
  1444.  -- SIP/201-00000063 is ringing
  1445.  
  1446. asteros*CLI>
  1447. 
  1448. <--- Transmitting (no NAT) to 192.168.127.138:64668 --->
  1449. SIP/2.0 180 Ringing
  1450. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-9e34ec0c7a7b965c-1--d87543-;received=192.168.127.138;rport=64668
  1451. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1452. To: "201"<sip:201@192.168.127.183>;tag=as0d27c444
  1453. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1454. CSeq: 2 INVITE
  1455. Server: Asterisk PBX 11.3.0
  1456. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1457. Supported: replaces, timer
  1458. Contact: <sip:201@192.168.127.183:5060>
  1459. Content-Length: 0
  1460.  
  1461.  
  1462. <------------>
  1463.  
  1464. asteros*CLI>
  1465. 
  1466. <--- SIP read from UDP:192.168.127.102:4134 --->
  1467. SIP/2.0 200 OK
  1468. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK3be07969
  1469. Contact: <sip:201@192.168.127.102:4134>
  1470. To: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>;tag=ececafca
  1471. From: "Eduardo Salazar"<sip:202@192.168.127.183>;tag=as78bcc2d1
  1472. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1473. CSeq: 102 INVITE
  1474. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  1475. Content-Type: application/sdp
  1476. Supported: replaces
  1477. User-Agent: X-Lite release 4.5 stamp 69607
  1478. Content-Length: 217
  1479.  
  1480. v=0
  1481. o=- 13012520628557962 3 IN IP4 192.168.127.102
  1482. s=X-Lite 4 release 4.5 stamp 69607
  1483. c=IN IP4 192.168.127.102
  1484. t=0 0
  1485. m=audio 56950 RTP/AVP 0 8 101
  1486. a=rtpmap:101 telephone-event/8000
  1487. a=fmtp:101 0-15
  1488. a=sendrecv
  1489. <------------->
  1490. --- (12 headers 9 lines) ---
  1491. Found RTP audio format 0
  1492. Found RTP audio format 8
  1493. Found RTP audio format 101
  1494. Found audio description format telephone-event for ID 101
  1495. Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  1496. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  1497. Peer audio RTP is at port 192.168.127.102:56950
  1498. list_route: hop: <sip:201@192.168.127.102:4134>
  1499. set_destination: Parsing <sip:201@192.168.127.102:4134> for address/port to send to
  1500. set_destination: set destination to 192.168.127.102:4134
  1501. Transmitting (no NAT) to 192.168.127.102:4134:
  1502. ACK sip:201@192.168.127.102:4134 SIP/2.0
  1503. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK6f1ef581
  1504. Max-Forwards: 70
  1505. From: "Eduardo Salazar" <sip:202@192.168.127.183>;tag=as78bcc2d1
  1506. To: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>;tag=ececafca
  1507. Contact: <sip:202@192.168.127.183:5060>
  1508. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1509. CSeq: 102 ACK
  1510. User-Agent: Asterisk PBX 11.3.0
  1511. Content-Length: 0
  1512.  
  1513.  
  1514. ---
  1515.  
  1516. asteros*CLI>
  1517.  -- SIP/201-00000063 answered SIP/202-00000062
  1518. Audio is at 16766
  1519. Adding codec 100003 (ulaw) to SDP
  1520. Adding codec 100004 (alaw) to SDP
  1521. Adding non-codec 0x1 (telephone-event) to SDP
  1522.  
  1523. <--- Reliably Transmitting (no NAT) to 192.168.127.138:64668 --->
  1524. SIP/2.0 200 OK
  1525. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-9e34ec0c7a7b965c-1--d87543-;received=192.168.127.138;rport=64668
  1526. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1527. To: "201"<sip:201@192.168.127.183>;tag=as0d27c444
  1528. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1529. CSeq: 2 INVITE
  1530. Server: Asterisk PBX 11.3.0
  1531. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1532. Supported: replaces, timer
  1533. Contact: <sip:201@192.168.127.183:5060>
  1534. Content-Type: application/sdp
  1535. Content-Length: 292
  1536.  
  1537. v=0
  1538. o=root 1441534548 1441534548 IN IP4 192.168.127.183
  1539. s=Asterisk PBX 11.3.0
  1540. c=IN IP4 192.168.127.183
  1541. t=0 0
  1542. m=audio 16766 RTP/AVP 0 8 101
  1543. a=rtpmap:0 PCMU/8000
  1544. a=rtpmap:8 PCMA/8000
  1545. a=rtpmap:101 telephone-event/8000
  1546. a=fmtp:101 0-16
  1547. a=silenceSupp:off - - - -
  1548. a=ptime:20
  1549. a=sendrecv
  1550.  
  1551. <------------>
  1552.  
  1553. asteros*CLI>
  1554. 
  1555. <--- SIP read from UDP:192.168.127.138:64668 --->
  1556. ACK sip:201@192.168.127.183:5060 SIP/2.0
  1557. Via: SIP/2.0/UDP 192.168.127.138:64668;branch=z9hG4bK-d87543-c63d3b20383fff54-1--d87543-;rport
  1558. Max-Forwards: 70
  1559. Contact: <sip:202@192.168.127.138:64668>
  1560. To: "201"<sip:201@192.168.127.183>;tag=as0d27c444
  1561. From: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1562. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1563. CSeq: 2 ACK
  1564. User-Agent: X-Lite release 1011s stamp 41150
  1565. Authorization: Digest username="202",realm="asterisk",nonce="7c084d9d",uri="sip:201@192.168.127.183",response="671262764f3f01fc02063a1510075361",algorithm=MD5
  1566. Content-Length: 0
  1567.  
  1568. <------------->
  1569. --- (11 headers 0 lines) ---
  1570.  
  1571. asteros*CLI>
  1572. Really destroying SIP dialog '323caa12109de0292075762b14c38ec9@192.168.127.183:5060' Method: BYE
  1573.  
  1574. asteros*CLI>
  1575. 
  1576. <--- SIP read from UDP:192.168.127.102:4134 --->
  1577. BYE sip:202@192.168.127.183:5060 SIP/2.0
  1578. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-023dbf20d45c008a-1---d8754z-;rport
  1579. Max-Forwards: 70
  1580. Contact: <sip:201@192.168.127.102:4134>
  1581. To: "Eduardo Salazar"<sip:202@192.168.127.183>;tag=as78bcc2d1
  1582. From: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>;tag=ececafca
  1583. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1584. CSeq: 2 BYE
  1585. User-Agent: X-Lite release 4.5 stamp 69607
  1586. Content-Length: 0
  1587.  
  1588. <------------->
  1589. --- (10 headers 0 lines) ---
  1590. Sending to 192.168.127.102:4134 (no NAT)
  1591. Scheduling destruction of SIP dialog '364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060' in 32000 ms (Method: BYE)
  1592.  
  1593. <--- Transmitting (no NAT) to 192.168.127.102:4134 --->
  1594. SIP/2.0 200 OK
  1595. Via: SIP/2.0/UDP 192.168.127.102:4134;branch=z9hG4bK-d8754z-023dbf20d45c008a-1---d8754z-;received=192.168.127.102;rport=4134
  1596. From: <sip:201@192.168.127.102:4134;rinstance=21a7fb413c1a5671>;tag=ececafca
  1597. To: "Eduardo Salazar"<sip:202@192.168.127.183>;tag=as78bcc2d1
  1598. Call-ID: 364d2b7918baa8c976c8886e0d86fadb@192.168.127.183:5060
  1599. CSeq: 2 BYE
  1600. Server: Asterisk PBX 11.3.0
  1601. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1602. Supported: replaces, timer
  1603. Content-Length: 0
  1604.  
  1605.  
  1606. <------------>
  1607.  
  1608. asteros*CLI>
  1609.  == Spawn extension (internal, 201, 1) exited non-zero on 'SIP/202-00000062'
  1610.  
  1611. asteros*CLI>
  1612. Scheduling destruction of SIP dialog 'NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.' in 32000 ms (Method: ACK)
  1613.  
  1614. asteros*CLI>
  1615. set_destination: Parsing <sip:202@192.168.127.138:64668> for address/port to send to
  1616.  
  1617. asteros*CLI>
  1618. set_destination: set destination to 192.168.127.138:64668
  1619.  
  1620. asteros*CLI>
  1621. Reliably Transmitting (no NAT) to 192.168.127.138:64668:
  1622. BYE sip:202@192.168.127.138:64668 SIP/2.0
  1623. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK65fd805e;rport
  1624. Max-Forwards: 70
  1625. From: "201"<sip:201@192.168.127.183>;tag=as0d27c444
  1626. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1627. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1628. CSeq: 102 BYE
  1629. User-Agent: Asterisk PBX 11.3.0
  1630. Proxy-Authorization: Digest username="202", realm="asterisk", algorithm=MD5, uri="sip:192.168.127.183", nonce="7c084d9d", response="888f27ac3796b7e48ff832b1a6209841"
  1631. X-Asterisk-HangupCause: Normal Clearing
  1632. X-Asterisk-HangupCauseCode: 16
  1633. Content-Length: 0
  1634.  
  1635.  
  1636. ---
  1637.  
  1638. asteros*CLI>
  1639. 
  1640. <--- SIP read from UDP:192.168.127.138:64668 --->
  1641. SIP/2.0 200 OK
  1642. Via: SIP/2.0/UDP 192.168.127.183:5060;branch=z9hG4bK65fd805e;rport=5060
  1643. Contact: <sip:202@192.168.127.138:64668>
  1644. To: "Brian Salazar"<sip:202@192.168.127.183>;tag=7c4b7700
  1645. From: "201"<sip:201@192.168.127.183>;tag=as0d27c444
  1646. Call-ID: NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.
  1647. CSeq: 102 BYE
  1648. User-Agent: X-Lite release 1011s stamp 41150
  1649. Content-Length: 0
  1650.  
  1651. <------------->
  1652. --- (9 headers 0 lines) ---
  1653. SIP Response message for INCOMING dialog BYE arrived
  1654. Really destroying SIP dialog 'NmY3NmIzNzkxMTA3MTU0OGU5ZWY2NGFlZDQ0ODFjMTM.' Method: ACK
  1655.  
  1656. asteros*CLI>
  1657. 
  1658. <--- SIP read from UDP:192.168.127.102:4134 --->
  1659.  
  1660.  
  1661. <------------->
  1662.  
  1663. asteros*CLI>
  1664. Really destroying SIP dialog 'NjIxYjE0ZTE3ZGY1NTIxYWVlNzVjZjNhYTZlZTE4MTQ.' Method: SUBSCRIBE
  1665.  
  1666. asteros*CLI>
  1667. 
  1668. <--- SIP read from UDP:192.168.127.108:7572 --->
  1669.  
  1670.  
  1671. <------------->
  1672.  
  1673. asteros*CLI>
  1674. 
  1675. <--- SIP read from UDP:192.168.127.138:64668 --->
  1676.  
  1677.  
  1678. <------------->
  1679.  
  1680. asteros*CLI> quit
  1681. Asterisk cleanly ending (0).
  1682. Executing last minute cleanups
  1683. Asterisk ending (0).
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