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  1. Script started on Fri 06 Feb 2015 07:58:08 PM VLAT
  2. ]0;root@localhost:~[?1034h[root@localhost ~]# rebootstartservice sshd restart./call.shmc
  3. [root@localhost ~]# mcasterisk -vvvvvr
  4. Asterisk 11.14.1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  5. Created by Mark Spencer <markster@digium.com>
  6. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  7. This is free software, with components licensed under the GNU General Public
  8. License version 2 and other licenses; you are welcome to redistribute it under
  9. certain conditions. Type 'core show license' for details.
  10. =========================================================================
  11. Connected to Asterisk 11.14.1 currently running on localhost (pid = 1692)
  12. localhost*CLI> core set verbose 3sip set debug peer Trunk_Alcatelcore set verbose 3
  13.  
  14. localhost*CLI>
  15. Console verbose was 5 and is now 3.
  16.  
  17. localhost*CLI> core set verbose 3sip set debug peer Trunk_Alcatel
  18.  
  19. localhost*CLI>
  20. SIP Debugging Enabled for IP: 10.65.9.4
  21.  
  22. localhost*CLI>
  23. [2015-02-06 19:59:43] WARNING[1741]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/23486156504648.call: Operation not permitted
  24.  
  25. localhost*CLI>
  26.     -- Attempting call on SIP/Trunk_Alcatel/89004267333 for application Playback(demo-congrats) (Retry 1)
  27.  
  28. localhost*CLI>
  29.   == Using SIP RTP TOS bits 184
  30.  
  31. localhost*CLI>
  32.   == Using SIP RTP CoS mark 5
  33.  
  34. localhost*CLI>
  35. Audio is at 17606
  36.  
  37. localhost*CLI>
  38. Adding codec 100004 (alaw) to SDP
  39.  
  40. localhost*CLI>
  41. Adding codec 100003 (ulaw) to SDP
  42.  
  43. localhost*CLI>
  44. Adding non-codec 0x1 (telephone-event) to SDP
  45.  
  46. localhost*CLI>
  47. Reliably Transmitting (no NAT) to 10.65.9.4:5060:
  48. INVITE sip:89004267333@10.65.9.4 SIP/2.0
  49.  
  50. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
  51.  
  52. Max-Forwards: 70
  53.  
  54. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
  55.  
  56. To: <sip:89004267333@10.65.9.4>
  57.  
  58. Contact: <sip:Unknown@10.65.14.247:5060>
  59.  
  60. Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
  61.  
  62. CSeq: 102 INVITE
  63.  
  64. User-Agent: FPBX-2.11.0(11.14.1)
  65.  
  66. Date: Fri, 06 Feb 2015 09:59:43 GMT
  67.  
  68. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  69.  
  70. Supported: replaces, timer
  71.  
  72. Content-Type: application/sdp
  73.  
  74. Content-Length: 258
  75.  
  76.  
  77.  
  78. v=0
  79.  
  80. o=root 770304551 770304551 IN IP4 10.65.14.247
  81.  
  82. s=Asterisk PBX 11.14.1
  83.  
  84. c=IN IP4 10.65.14.247
  85.  
  86. t=0 0
  87.  
  88. m=audio 17606 RTP/AVP 8 0 101
  89.  
  90. a=rtpmap:8 PCMA/8000
  91.  
  92. a=rtpmap:0 PCMU/8000
  93.  
  94. a=rtpmap:101 telephone-event/8000
  95.  
  96. a=fmtp:101 0-16
  97.  
  98. a=ptime:20
  99.  
  100. a=sendrecv
  101.  
  102.  
  103. ---
  104.  
  105. localhost*CLI>
  106. 
  107. <--- SIP read from UDP:10.65.9.4:5060 --->
  108. SIP/2.0 100 Trying
  109. To: <sip:89004267333@10.65.9.4>
  110. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
  111. Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
  112. CSeq: 102 INVITE
  113. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
  114. Content-Length: 0
  115.  
  116. <------------->
  117. --- (7 headers 0 lines) ---
  118.  
  119. localhost*CLI>
  120. 
  121. <--- SIP read from UDP:10.65.9.4:5060 --->
  122. SIP/2.0 502 Bad Gateway
  123. User-Agent: OmniPCX Enterprise R9.1 i1.605.29
  124. To: <sip:89004267333@10.65.9.4>;tag=e1ba17177d1cb58cdb44a959dc91d967
  125. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
  126. Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
  127. CSeq: 102 INVITE
  128. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
  129. Content-Length: 0
  130.  
  131. <------------->
  132. --- (8 headers 0 lines) ---
  133.     -- Got SIP response 502 "Bad Gateway" back from 10.65.9.4:5060
  134. Transmitting (no NAT) to 10.65.9.4:5060:
  135. ACK sip:89004267333@10.65.9.4 SIP/2.0
  136.  
  137. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK7d4b6f77
  138.  
  139. Max-Forwards: 70
  140.  
  141. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as404e1978
  142.  
  143. To: <sip:89004267333@10.65.9.4>;tag=e1ba17177d1cb58cdb44a959dc91d967
  144.  
  145. Contact: <sip:Unknown@10.65.14.247:5060>
  146.  
  147. Call-ID: 203c4ba9352392994569cc684d18e03a@10.65.14.247:5060
  148.  
  149. CSeq: 102 ACK
  150.  
  151. User-Agent: FPBX-2.11.0(11.14.1)
  152.  
  153. Content-Length: 0
  154.  
  155.  
  156.  
  157.  
  158. ---
  159.  
  160. localhost*CLI>
  161. [2015-02-06 19:59:44] NOTICE[2400]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
  162. [2015-02-06 19:59:44] NOTICE[2400]: pbx_spool.c:392 attempt_thread: Queued call to SIP/Trunk_Alcatel/89004267333 expired without completion after 0 attempts
  163.  
  164. localhost*CLI>
  165. Really destroying SIP dialog '203c4ba9352392994569cc684d18e03a@10.65.14.247:5060' Method: INVITE
  166.  
  167. localhost*CLI>
  168. Reliably Transmitting (no NAT) to 10.65.9.4:5060:
  169. OPTIONS sip:10.65.9.4 SIP/2.0
  170.  
  171. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK2690a3ec
  172.  
  173. Max-Forwards: 70
  174.  
  175. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as0256152c
  176.  
  177. To: <sip:10.65.9.4>
  178.  
  179. Contact: <sip:Unknown@10.65.14.247:5060>
  180.  
  181. Call-ID: 632632145529579d507bcc6e4402f852@10.65.14.247:5060
  182.  
  183. CSeq: 102 OPTIONS
  184.  
  185. User-Agent: FPBX-2.11.0(11.14.1)
  186.  
  187. Date: Fri, 06 Feb 2015 10:00:13 GMT
  188.  
  189. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  190.  
  191. Supported: replaces, timer
  192.  
  193. Content-Length: 0
  194.  
  195.  
  196.  
  197.  
  198. ---
  199.  
  200. localhost*CLI>
  201. 
  202. <--- SIP read from UDP:10.65.9.4:5060 --->
  203. SIP/2.0 200 OK
  204. Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
  205. Contact: sip:10.65.9.4
  206. Supported: replaces,timer,100rel
  207. User-Agent: OmniPCX Enterprise R9.1 i1.605.29
  208. To: <sip:10.65.9.4>;tag=39e64c72a49c140608a50df2ba47e693
  209. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as0256152c
  210. Call-ID: 632632145529579d507bcc6e4402f852@10.65.14.247:5060
  211. CSeq: 102 OPTIONS
  212. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK2690a3ec
  213. Content-Length: 0
  214.  
  215. <------------->
  216. --- (11 headers 0 lines) ---
  217. Really destroying SIP dialog '632632145529579d507bcc6e4402f852@10.65.14.247:5060' Method: OPTIONS
  218.  
  219. localhost*CLI>
  220. Reliably Transmitting (no NAT) to 10.65.9.4:5060:
  221. OPTIONS sip:10.65.9.4 SIP/2.0
  222.  
  223. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK15392c1c
  224.  
  225. Max-Forwards: 70
  226.  
  227. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as41c48d9c
  228.  
  229. To: <sip:10.65.9.4>
  230.  
  231. Contact: <sip:Unknown@10.65.14.247:5060>
  232.  
  233. Call-ID: 37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060
  234.  
  235. CSeq: 102 OPTIONS
  236.  
  237. User-Agent: FPBX-2.11.0(11.14.1)
  238.  
  239. Date: Fri, 06 Feb 2015 10:01:13 GMT
  240.  
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  242.  
  243. Supported: replaces, timer
  244.  
  245. Content-Length: 0
  246.  
  247.  
  248.  
  249.  
  250. ---
  251.  
  252. localhost*CLI>
  253. 
  254. <--- SIP read from UDP:10.65.9.4:5060 --->
  255. SIP/2.0 200 OK
  256. Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
  257. Contact: sip:10.65.9.4
  258. Supported: replaces,timer,100rel
  259. User-Agent: OmniPCX Enterprise R9.1 i1.605.29
  260. To: <sip:10.65.9.4>;tag=f79fdf1fa8fdbb563747edac2ba5dfcc
  261. From: "Unknown" <sip:Unknown@10.65.14.247>;tag=as41c48d9c
  262. Call-ID: 37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060
  263. CSeq: 102 OPTIONS
  264. Via: SIP/2.0/UDP 10.65.14.247:5060;branch=z9hG4bK15392c1c
  265. Content-Length: 0
  266.  
  267. <------------->
  268. --- (11 headers 0 lines) ---
  269. Really destroying SIP dialog '37205e9e5ccb0b4405a661ef119a1290@10.65.14.247:5060' Method: OPTIONS
  270.  
  271. localhost*CLI>
  272. Disconnected from Asterisk server
  273. Asterisk cleanly ending (0).
  274. Executing last minute cleanups
  275. ]0;root@localhost:~[root@localhost ~]# asterisk -vvvvvrrebootstartservice sshd restart./call.shmc./call.shservice sshd restartrestartbootasterisk -vvvvvrasterisk -vvvvvrexit
  276. exit
  277.  
  278. Script done on Fri 06 Feb 2015 08:01:47 PM VLAT
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