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  1. sip reload
  2. =================================================================================
  3. asterisk*CLI>
  4. asterisk*CLI>
  5. asterisk*CLI> sip reload
  6. Reloading SIP
  7. == Parsing '/etc/asterisk/sip.conf': == Found
  8. == Parsing '/etc/asterisk/sip_general_additional.conf': == Found
  9. == Parsing '/etc/asterisk/sip_general_custom.conf': == Found
  10. == Parsing '/etc/asterisk/sip_nat.conf': == Found
  11. == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found
  12. == Parsing '/etc/asterisk/sip_registrations.conf': == Found
  13. == Parsing '/etc/asterisk/sip_custom.conf': == Found
  14. == Parsing '/etc/asterisk/sip_additional.conf': == Found
  15. == Parsing '/etc/asterisk/sip_custom_post.conf': == Found
  16. == Parsing '/etc/asterisk/users.conf': == Found
  17. == Using SIP TOS bits 96
  18. == Using SIP CoS mark 4
  19. Reliably Transmitting (NAT) to 192.168.6.14:5060:
  20. OPTIONS sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b SIP/2.0
  21. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5e39f50a;rport
  22. Max-Forwards: 70
  23. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as490fa925
  24. To: <sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b>
  25. Contact: <sip:Unknown@192.168.6.16:5060>
  26. Call-ID: 5925ce7d272dd01008ace743739c1b63@192.168.6.16:5060
  27. CSeq: 102 OPTIONS
  28. User-Agent: FPBX-2.8.1(1.8.7.0)
  29. Date: Sun, 16 Sep 2012 08:47:21 GMT
  30. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  31. Supported: replaces, timer
  32. Content-Length: 0
  33.  
  34.  
  35. ---
  36. Scheduling destruction of SIP dialog '0c21b579182d502079ca4d1d54f61b09@192.168.6.16:5060' in 17216 ms (Method: NOTIFY)
  37. Reliably Transmitting (NAT) to 192.168.6.14:5060:
  38. NOTIFY sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b SIP/2.0
  39. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK597188fa;rport
  40. Max-Forwards: 70
  41. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as754b1622
  42. To: <sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b>
  43. Contact: <sip:Unknown@192.168.6.16:5060>
  44. Call-ID: 0c21b579182d502079ca4d1d54f61b09@192.168.6.16:5060
  45. CSeq: 102 NOTIFY
  46. User-Agent: FPBX-2.8.1(1.8.7.0)
  47. Event: message-summary
  48. Content-Type: application/simple-message-summary
  49. Content-Length: 92
  50.  
  51. Messages-Waiting: no
  52. Message-Account: sip:*97@192.168.6.16:5060
  53. Voice-Message: 0/0 (0/0)
  54.  
  55. ---
  56. == Parsing '/etc/asterisk/sip_notify.conf': == Found
  57. == Parsing '/etc/asterisk/sip_notify_custom.conf': == Found
  58. == Parsing '/etc/asterisk/sip_notify_custom_elastix.conf': == Found
  59. == Parsing '/etc/asterisk/sip_notify_additional.conf': == Found
  60.  
  61. <--- SIP read from UDP:192.168.6.14:5060 --->
  62. SIP/2.0 200 OK
  63. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5e39f50a;rport=5060
  64. Contact: <sip:10.0.0.2:5060>
  65. To: <sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b>;tag=0e60d97e
  66. From: "Unknown"<sip:Unknown@192.168.6.16>;tag=as490fa925
  67. Call-ID: 5925ce7d272dd01008ace743739c1b63@192.168.6.16:5060
  68. CSeq: 102 OPTIONS
  69. Accept: application/sdp
  70. Accept-Language: en
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  72. Supported: replaces
  73. User-Agent: X-Lite release 5.0.0 stamp 67284
  74. Content-Length: 0
  75.  
  76. <------------->
  77. --- (13 headers 0 lines) ---
  78. Reliably Transmitting (NAT) to 109.68.162.233:5060:
  79. OPTIONS sip:sip.mydivert.com SIP/2.0
  80. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK0e044033;rport
  81. Max-Forwards: 70
  82. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as7b6bd885
  83. To: <sip:sip.mydivert.com>
  84. Contact: <sip:Unknown@192.168.6.16:5060>
  85. Call-ID: 42ec4b717e8d079a00311da65396fdaf@192.168.6.16:5060
  86. CSeq: 102 OPTIONS
  87. User-Agent: FPBX-2.8.1(1.8.7.0)
  88. Date: Sun, 16 Sep 2012 08:47:22 GMT
  89. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  90. Supported: replaces, timer
  91. Content-Length: 0
  92.  
  93.  
  94. ---
  95. Really destroying SIP dialog '5925ce7d272dd01008ace743739c1b63@192.168.6.16:5060' Method: OPTIONS
  96.  
  97. <--- SIP read from UDP:192.168.6.14:5060 --->
  98. SIP/2.0 200 OK
  99. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK597188fa;rport=5060
  100. Contact: <sip:10.0.0.2:5060>
  101. To: <sip:4002@10.0.0.2:5060;rinstance=dfd4b0d2b95d4f2b>;tag=968381fa
  102. From: "Unknown"<sip:Unknown@192.168.6.16>;tag=as754b1622
  103. Call-ID: 0c21b579182d502079ca4d1d54f61b09@192.168.6.16:5060
  104. CSeq: 102 NOTIFY
  105. User-Agent: X-Lite release 5.0.0 stamp 67284
  106. Content-Length: 0
  107.  
  108. <------------->
  109. --- (9 headers 0 lines) ---
  110. Really destroying SIP dialog '0c21b579182d502079ca4d1d54f61b09@192.168.6.16:5060' Method: NOTIFY
  111.  
  112. <--- SIP read from UDP:109.68.162.233:5060 --->
  113. SIP/2.0 200 OK
  114. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK0e044033;rport=1181
  115. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as7b6bd885
  116. To: <sip:sip.mydivert.com>;tag=af2122e3c448f11e51eed26b8203debd.268b
  117. Call-ID: 42ec4b717e8d079a00311da65396fdaf@192.168.6.16:5060
  118. CSeq: 102 OPTIONS
  119. Server: VoipNow
  120. Content-Length: 0
  121.  
  122. <------------->
  123. --- (8 headers 0 lines) ---
  124. Really destroying SIP dialog '42ec4b717e8d079a00311da65396fdaf@192.168.6.16:5060' Method: OPTIONS
  125. asterisk*CLI>
  126.  
  127.  
  128.  
  129. sip show peers
  130. ============================================================
  131. asterisk*CLI> sip show peers
  132. Name/username Host Dyn Forcerport ACL Port Status
  133. 4001 (Unspecified) D N A 0 UNKNOWN
  134. 4002/4002 192.168.6.14 D N A 5060 OK (4 ms)
  135. mydivert-out/billing@thin 109.68.162.233 N 5060 OK (233 ms)
  136. 3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]
  137. asterisk*CLI>
  138.  
  139.  
  140.  
  141.  
  142.  
  143.  
  144.  
  145.  
  146.  
  147.  
  148. calling to 94714887904
  149. ============================================================
  150. asterisk*CLI>
  151. asterisk*CLI>
  152. asterisk*CLI> sip set debug off
  153. SIP Debugging Disabled
  154. asterisk*CLI> sip set debug on
  155. SIP Debugging enabled
  156. asterisk*CLI>
  157. asterisk*CLI>
  158. Reliably Transmitting (NAT) to 109.68.162.233:5060:
  159. OPTIONS sip:sip.mydivert.com SIP/2.0
  160. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK1b49a4b3;rport
  161. Max-Forwards: 70
  162. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as2c13bdad
  163. To: <sip:sip.mydivert.com>
  164. Contact: <sip:Unknown@192.168.6.16:5060>
  165. Call-ID: 5b7f3178021bdf305d9b5cc5214a04b4@192.168.6.16:5060
  166. CSeq: 102 OPTIONS
  167. User-Agent: FPBX-2.8.1(1.8.7.0)
  168. Date: Sun, 16 Sep 2012 08:26:00 GMT
  169. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  170. Supported: replaces, timer
  171. Content-Length: 0
  172.  
  173.  
  174. ---
  175.  
  176. <--- SIP read from UDP:109.68.162.233:5060 --->
  177. SIP/2.0 200 OK
  178. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK1b49a4b3;rport=1181
  179. From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as2c13bdad
  180. To: <sip:sip.mydivert.com>;tag=af2122e3c448f11e51eed26b8203debd.62b6
  181. Call-ID: 5b7f3178021bdf305d9b5cc5214a04b4@192.168.6.16:5060
  182. CSeq: 102 OPTIONS
  183. Server: VoipNow
  184. Content-Length: 0
  185.  
  186. <------------->
  187. --- (8 headers 0 lines) ---
  188. Really destroying SIP dialog '5b7f3178021bdf305d9b5cc5214a04b4@192.168.6.16:5060' Method: OPTIONS
  189.  
  190. <--- SIP read from UDP:192.168.6.14:5060 --->
  191.  
  192.  
  193. <------------->
  194.  
  195. <--- SIP read from UDP:192.168.6.14:5060 --->
  196. INVITE sip:94714887904@192.168.6.16 SIP/2.0
  197. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-0beb2c01e46b2fd5-1---d8754z-;rport
  198. Max-Forwards: 70
  199. Contact: <sip:4002@10.0.0.2:5060>
  200. To: <sip:94714887904@192.168.6.16>
  201. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  202. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  203. CSeq: 1 INVITE
  204. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  205. Content-Type: application/sdp
  206. Supported: replaces
  207. User-Agent: X-Lite release 5.0.0 stamp 67284
  208. Content-Length: 234
  209.  
  210. v=0
  211. o=- 12992257579877184 1 IN IP4 10.0.0.2
  212. s=CounterPath X-Lite 5.0.0
  213. c=IN IP4 10.0.0.2
  214. b=AS:1638
  215. t=0 0
  216. m=audio 5062 RTP/AVP 107 0 8 101
  217. a=rtpmap:107 BV32/16000
  218. a=rtpmap:101 telephone-event/8000
  219. a=fmtp:101 0-15
  220. a=sendrecv
  221. <------------->
  222. --- (13 headers 11 lines) ---
  223. Sending to 192.168.6.14:5060 (no NAT)
  224. Using INVITE request as basis request - OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  225. Found peer '4002' for '4002' from 192.168.6.14:5060
  226.  
  227. <--- Reliably Transmitting (NAT) to 192.168.6.14:5060 --->
  228. SIP/2.0 401 Unauthorized
  229. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-0beb2c01e46b2fd5-1---d8754z-;received=192.168.6.14;rport=5060
  230. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  231. To: <sip:94714887904@192.168.6.16>;tag=as1b58eea2
  232. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  233. CSeq: 1 INVITE
  234. Server: FPBX-2.8.1(1.8.7.0)
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  236. Supported: replaces, timer
  237. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23db7b98"
  238. Content-Length: 0
  239.  
  240.  
  241. <------------>
  242. Scheduling destruction of SIP dialog 'OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.' in 6400 ms (Method: INVITE)
  243.  
  244. <--- SIP read from UDP:192.168.6.14:5060 --->
  245. ACK sip:94714887904@192.168.6.16 SIP/2.0
  246. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-0beb2c01e46b2fd5-1---d8754z-;rport
  247. Max-Forwards: 70
  248. To: <sip:94714887904@192.168.6.16>;tag=as1b58eea2
  249. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  250. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  251. CSeq: 1 ACK
  252. Content-Length: 0
  253.  
  254. <------------->
  255. --- (8 headers 0 lines) ---
  256.  
  257. <--- SIP read from UDP:192.168.6.14:5060 --->
  258. INVITE sip:94714887904@192.168.6.16 SIP/2.0
  259. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-73f859fe680c275e-1---d8754z-;rport
  260. Max-Forwards: 70
  261. Contact: <sip:4002@10.0.0.2:5060>
  262. To: <sip:94714887904@192.168.6.16>
  263. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  264. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  265. CSeq: 2 INVITE
  266. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  267. Content-Type: application/sdp
  268. Supported: replaces
  269. User-Agent: X-Lite release 5.0.0 stamp 67284
  270. Authorization: Digest username="4002",realm="asterisk",nonce="23db7b98",uri="sip:94714887904@192.168.6.16",response="953285afb41fd5f15025155df283efeb",algorithm=MD5
  271. Content-Length: 234
  272.  
  273. v=0
  274. o=- 12992257579877184 1 IN IP4 10.0.0.2
  275. s=CounterPath X-Lite 5.0.0
  276. c=IN IP4 10.0.0.2
  277. b=AS:1638
  278. t=0 0
  279. m=audio 5062 RTP/AVP 107 0 8 101
  280. a=rtpmap:107 BV32/16000
  281. a=rtpmap:101 telephone-event/8000
  282. a=fmtp:101 0-15
  283. a=sendrecv
  284. <------------->
  285. --- (14 headers 11 lines) ---
  286. Sending to 192.168.6.14:5060 (NAT)
  287. Using INVITE request as basis request - OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  288. Found peer '4002' for '4002' from 192.168.6.14:5060
  289. == Using SIP RTP TOS bits 184
  290. == Using SIP RTP CoS mark 5
  291. Found RTP audio format 107
  292. Found RTP audio format 0
  293. Found RTP audio format 8
  294. Found RTP audio format 101
  295. Found unknown media description format BV32 for ID 107
  296. Found audio description format telephone-event for ID 101
  297. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  298. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  299. Peer audio RTP is at port 10.0.0.2:5062
  300. Looking for 94714887904 in from-internal (domain 192.168.6.16)
  301. list_route: hop: <sip:4002@10.0.0.2:5060>
  302.  
  303. <--- Transmitting (NAT) to 192.168.6.14:5060 --->
  304. SIP/2.0 100 Trying
  305. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-73f859fe680c275e-1---d8754z-;received=192.168.6.14;rport=5060
  306. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  307. To: <sip:94714887904@192.168.6.16>
  308. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  309. CSeq: 2 INVITE
  310. Server: FPBX-2.8.1(1.8.7.0)
  311. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  312. Supported: replaces, timer
  313. Contact: <sip:94714887904@192.168.6.16:5060>
  314. Content-Length: 0
  315.  
  316.  
  317. <------------>
  318. -- Executing [94714887904@from-internal:1] Macro("SIP/4002-0000001d", "user-callerid,SKIPTTL,") in new stack
  319. -- Executing [s@macro-user-callerid:1] Set("SIP/4002-0000001d", "AMPUSER=4002") in new stack
  320. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4002-0000001d", "0?report") in new stack
  321. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4002-0000001d", "1?Set(REALCALLERIDNUM=4002)") in new stack
  322. -- Executing [s@macro-user-callerid:4] Set("SIP/4002-0000001d", "AMPUSER=4002") in new stack
  323. -- Executing [s@macro-user-callerid:5] Set("SIP/4002-0000001d", "AMPUSERCIDNAME=4002") in new stack
  324. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4002-0000001d", "0?report") in new stack
  325. -- Executing [s@macro-user-callerid:7] Set("SIP/4002-0000001d", "AMPUSERCID=4002") in new stack
  326. -- Executing [s@macro-user-callerid:8] Set("SIP/4002-0000001d", "CALLERID(all)="4002" <4002>") in new stack
  327. -- Executing [s@macro-user-callerid:9] ExecIf("SIP/4002-0000001d", "0?Set(CHANNEL(language)=)") in new stack
  328. -- Executing [s@macro-user-callerid:10] GotoIf("SIP/4002-0000001d", "1?continue") in new stack
  329. -- Goto (macro-user-callerid,s,19)
  330. -- Executing [s@macro-user-callerid:19] Set("SIP/4002-0000001d", "CALLERID(number)=4002") in new stack
  331. -- Executing [s@macro-user-callerid:20] Set("SIP/4002-0000001d", "CALLERID(name)=4002") in new stack
  332. -- Executing [s@macro-user-callerid:21] NoOp("SIP/4002-0000001d", "Using CallerID "4002" <4002>") in new stack
  333. -- Executing [94714887904@from-internal:2] NoOp("SIP/4002-0000001d", "Calling Out Route: to-mydivert") in new stack
  334. -- Executing [94714887904@from-internal:3] Set("SIP/4002-0000001d", "MOHCLASS=default") in new stack
  335. -- Executing [94714887904@from-internal:4] Set("SIP/4002-0000001d", "_NODEST=") in new stack
  336. -- Executing [94714887904@from-internal:5] Macro("SIP/4002-0000001d", "record-enable,4002,OUT,") in new stack
  337. -- Executing [s@macro-record-enable:1] GotoIf("SIP/4002-0000001d", "1?check") in new stack
  338. -- Goto (macro-record-enable,s,4)
  339. -- Executing [s@macro-record-enable:4] ExecIf("SIP/4002-0000001d", "0?MacroExit()") in new stack
  340. -- Executing [s@macro-record-enable:5] GotoIf("SIP/4002-0000001d", "0?Group:OUT") in new stack
  341. -- Goto (macro-record-enable,s,15)
  342. -- Executing [s@macro-record-enable:15] GotoIf("SIP/4002-0000001d", "0?IN") in new stack
  343. -- Executing [s@macro-record-enable:16] ExecIf("SIP/4002-0000001d", "1?MacroExit()") in new stack
  344. -- Executing [94714887904@from-internal:6] Macro("SIP/4002-0000001d", "dialout-trunk,2,94714887904,") in new stack
  345. -- Executing [s@macro-dialout-trunk:1] Set("SIP/4002-0000001d", "DIAL_TRUNK=2") in new stack
  346. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4002-0000001d", "0?sub-pincheck,s,1") in new stack
  347. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4002-0000001d", "0?disabletrunk,1") in new stack
  348. -- Executing [s@macro-dialout-trunk:4] Set("SIP/4002-0000001d", "DIAL_NUMBER=94714887904") in new stack
  349. -- Executing [s@macro-dialout-trunk:5] Set("SIP/4002-0000001d", "DIAL_TRUNK_OPTIONS=tr") in new stack
  350. -- Executing [s@macro-dialout-trunk:6] Set("SIP/4002-0000001d", "OUTBOUND_GROUP=OUT_2") in new stack
  351. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4002-0000001d", "1?nomax") in new stack
  352. -- Goto (macro-dialout-trunk,s,9)
  353. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4002-0000001d", "0?skipoutcid") in new stack
  354. -- Executing [s@macro-dialout-trunk:10] Set("SIP/4002-0000001d", "DIAL_TRUNK_OPTIONS=") in new stack
  355. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/4002-0000001d", "outbound-callerid,2") in new stack
  356. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/4002-0000001d", "0?Set(CALLERPRES()=)") in new stack
  357. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/4002-0000001d", "0?Set(REALCALLERIDNUM=4002)") in new stack
  358. -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/4002-0000001d", "1?normcid") in new stack
  359. -- Goto (macro-outbound-callerid,s,6)
  360. -- Executing [s@macro-outbound-callerid:6] Set("SIP/4002-0000001d", "USEROUTCID=4002") in new stack
  361. -- Executing [s@macro-outbound-callerid:7] Set("SIP/4002-0000001d", "EMERGENCYCID=") in new stack
  362. -- Executing [s@macro-outbound-callerid:8] Set("SIP/4002-0000001d", "TRUNKOUTCID="out"<9611361319>.") in new stack
  363. -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/4002-0000001d", "1?trunkcid") in new stack
  364. -- Goto (macro-outbound-callerid,s,12)
  365. -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/4002-0000001d", "1?Set(CALLERID(all)="out"<9611361319>.)") in new stack
  366. -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/4002-0000001d", "1?Set(CALLERID(all)=4002)") in new stack
  367. -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/4002-0000001d", "1?Set(CALLERID(all)="out"<9611361319>.)") in new stack
  368. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/4002-0000001d", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  369. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4002-0000001d", "1?sub-flp-2,s,1") in new stack
  370. -- Executing [s@sub-flp-2:1] ExecIf("SIP/4002-0000001d", "1?Return()") in new stack
  371. -- Executing [s@macro-dialout-trunk:13] Set("SIP/4002-0000001d", "OUTNUM=94714887904") in new stack
  372. -- Executing [s@macro-dialout-trunk:14] Set("SIP/4002-0000001d", "custom=SIP/mydivert-out") in new stack
  373. -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4002-0000001d", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
  374. -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4002-0000001d", "dialout-trunk-predial-hook,") in new stack
  375. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4002-0000001d", "") in new stack
  376. -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4002-0000001d", "0?bypass,1") in new stack
  377. -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4002-0000001d", "0?customtrunk") in new stack
  378. -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4002-0000001d", "SIP/mydivert-out/94714887904,300,") in new stack
  379. == Using SIP RTP TOS bits 184
  380. == Using SIP RTP CoS mark 5
  381. Audio is at 5060
  382. Adding codec 0x4 (ulaw) to SDP
  383. Adding codec 0x8 (alaw) to SDP
  384. Adding non-codec 0x1 (telephone-event) to SDP
  385. Reliably Transmitting (NAT) to 109.68.162.233:5060:
  386. INVITE sip:94714887904@sip.mydivert.com SIP/2.0
  387. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK3bb0c657;rport
  388. Max-Forwards: 70
  389. From: "out" <sip:billing@thinkmedialabs.com@192.168.6.16>;tag=as6e95b9c7
  390. To: <sip:94714887904@sip.mydivert.com>
  391. Contact: <sip:billing@thinkmedialabs.com@192.168.6.16:5060>
  392. Call-ID: 4c79754b569d2a702500c74e2c01177c@192.168.6.16:5060
  393. CSeq: 102 INVITE
  394. User-Agent: FPBX-2.8.1(1.8.7.0)
  395. Date: Sun, 16 Sep 2012 08:26:05 GMT
  396. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  397. Supported: replaces, timer
  398. Remote-Party-ID: "out" <sip:9611361319@192.168.6.16>;party=calling;privacy=off;screen=no
  399. Content-Type: application/sdp
  400. Content-Length: 260
  401.  
  402. v=0
  403. o=root 1173929662 1173929662 IN IP4 192.168.6.16
  404. s=Asterisk PBX 1.8.7.0
  405. c=IN IP4 192.168.6.16
  406. t=0 0
  407. m=audio 19342 RTP/AVP 0 8 101
  408. a=rtpmap:0 PCMU/8000
  409. a=rtpmap:8 PCMA/8000
  410. a=rtpmap:101 telephone-event/8000
  411. a=fmtp:101 0-16
  412. a=ptime:20
  413. a=sendrecv
  414.  
  415. ---
  416. -- Called SIP/mydivert-out/94714887904
  417.  
  418. <--- SIP read from UDP:109.68.162.233:5060 --->
  419. SIP/2.0 400 Bad Request
  420. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK3bb0c657;rport=1181;received=213.204.79.112
  421. From: "out" <sip:billing@thinkmedialabs.com@192.168.6.16>;tag=as6e95b9c7
  422. To: <sip:94714887904@sip.mydivert.com>;tag=af2122e3c448f11e51eed26b8203debd.3ad3
  423. Call-ID: 4c79754b569d2a702500c74e2c01177c@192.168.6.16:5060
  424. CSeq: 102 INVITE
  425. Server: VoipNow
  426. Content-Length: 0
  427.  
  428. <------------->
  429. --- (8 headers 0 lines) ---
  430. -- Got SIP response 400 "Bad Request" back from 109.68.162.233:5060
  431. Transmitting (NAT) to 109.68.162.233:5060:
  432. ACK sip:94714887904@sip.mydivert.com SIP/2.0
  433. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK3bb0c657;rport
  434. Max-Forwards: 70
  435. From: "out" <sip:billing@thinkmedialabs.com@192.168.6.16>;tag=as6e95b9c7
  436. To: <sip:94714887904@sip.mydivert.com>;tag=af2122e3c448f11e51eed26b8203debd.3ad3
  437. Contact: <sip:billing@thinkmedialabs.com@192.168.6.16:5060>
  438. Call-ID: 4c79754b569d2a702500c74e2c01177c@192.168.6.16:5060
  439. CSeq: 102 ACK
  440. User-Agent: FPBX-2.8.1(1.8.7.0)
  441. Content-Length: 0
  442.  
  443.  
  444. ---
  445. -- SIP/mydivert-out-0000001e is circuit-busy
  446. == Everyone is busy/congested at this time (1:0/1/0)
  447. -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4002-0000001d", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 127") in new stack
  448. -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4002-0000001d", "s-CONGESTION,1") in new stack
  449. -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  450. -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/4002-0000001d", "RC=127") in new stack
  451. -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/4002-0000001d", "127,1") in new stack
  452. -- Goto (macro-dialout-trunk,127,1)
  453. -- Executing [127@macro-dialout-trunk:1] Goto("SIP/4002-0000001d", "continue,1") in new stack
  454. -- Goto (macro-dialout-trunk,continue,1)
  455. -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/4002-0000001d", "1?noreport") in new stack
  456. -- Goto (macro-dialout-trunk,continue,3)
  457. -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/4002-0000001d", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 127 - failing through to other trunks") in new stack
  458. -- Executing [continue@macro-dialout-trunk:4] Set("SIP/4002-0000001d", "CALLERID(number)=4002") in new stack
  459. -- Executing [94714887904@from-internal:7] Macro("SIP/4002-0000001d", "outisbusy,") in new stack
  460. -- Executing [s@macro-outisbusy:1] Progress("SIP/4002-0000001d", "") in new stack
  461. Audio is at 5060
  462. Adding codec 0x4 (ulaw) to SDP
  463. Adding codec 0x8 (alaw) to SDP
  464. Adding non-codec 0x1 (telephone-event) to SDP
  465.  
  466. <--- Transmitting (NAT) to 192.168.6.14:5060 --->
  467. SIP/2.0 183 Session Progress
  468. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-73f859fe680c275e-1---d8754z-;received=192.168.6.14;rport=5060
  469. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  470. To: <sip:94714887904@192.168.6.16>;tag=as6c11f5a0
  471. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  472. CSeq: 2 INVITE
  473. Server: FPBX-2.8.1(1.8.7.0)
  474. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  475. Supported: replaces, timer
  476. Contact: <sip:94714887904@192.168.6.16:5060>
  477. Content-Type: application/sdp
  478. Content-Length: 260
  479.  
  480. v=0
  481. o=root 2090301594 2090301594 IN IP4 192.168.6.16
  482. s=Asterisk PBX 1.8.7.0
  483. c=IN IP4 192.168.6.16
  484. t=0 0
  485. m=audio 11084 RTP/AVP 0 8 101
  486. a=rtpmap:0 PCMU/8000
  487. a=rtpmap:8 PCMA/8000
  488. a=rtpmap:101 telephone-event/8000
  489. a=fmtp:101 0-16
  490. a=ptime:20
  491. a=sendrecv
  492.  
  493. <------------>
  494. -- Executing [s@macro-outisbusy:2] GotoIf("SIP/4002-0000001d", "0?emergency,1") in new stack
  495. -- Executing [s@macro-outisbusy:3] GotoIf("SIP/4002-0000001d", "0?intracompany,1") in new stack
  496. -- Executing [s@macro-outisbusy:4] Playback("SIP/4002-0000001d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
  497. -- <SIP/4002-0000001d> Playing 'all-circuits-busy-now.gsm' (language 'en')
  498. Really destroying SIP dialog '4c79754b569d2a702500c74e2c01177c@192.168.6.16:5060' Method: INVITE
  499. -- <SIP/4002-0000001d> Playing 'pls-try-call-later.gsm' (language 'en')
  500. -- Executing [s@macro-outisbusy:5] Congestion("SIP/4002-0000001d", "20") in new stack
  501.  
  502. <--- Reliably Transmitting (NAT) to 192.168.6.14:5060 --->
  503. SIP/2.0 503 Service Unavailable
  504. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-73f859fe680c275e-1---d8754z-;received=192.168.6.14;rport=5060
  505. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  506. To: <sip:94714887904@192.168.6.16>;tag=as6c11f5a0
  507. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  508. CSeq: 2 INVITE
  509. Server: FPBX-2.8.1(1.8.7.0)
  510. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  511. Supported: replaces, timer
  512. X-Asterisk-HangupCause: Interworking, unspecified
  513. X-Asterisk-HangupCauseCode: 127
  514. Content-Length: 0
  515.  
  516.  
  517. <------------>
  518. == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/4002-0000001d' in macro 'outisbusy'
  519. == Spawn extension (from-internal, 94714887904, 7) exited non-zero on 'SIP/4002-0000001d'
  520. -- Executing [h@from-internal:1] Macro("SIP/4002-0000001d", "hangupcall") in new stack
  521. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4002-0000001d", "1?endmixmoncheck") in new stack
  522. -- Goto (macro-hangupcall,s,9)
  523. -- Executing [s@macro-hangupcall:9] NoOp("SIP/4002-0000001d", "End of MIXMON check") in new stack
  524. -- Executing [s@macro-hangupcall:10] GotoIf("SIP/4002-0000001d", "1?nomeetmemon") in new stack
  525. -- Goto (macro-hangupcall,s,15)
  526. -- Executing [s@macro-hangupcall:15] NoOp("SIP/4002-0000001d", "MEETME_RECORDINGFILE=") in new stack
  527. -- Executing [s@macro-hangupcall:16] GotoIf("SIP/4002-0000001d", "1?noautomon") in new stack
  528. -- Goto (macro-hangupcall,s,18)
  529. -- Executing [s@macro-hangupcall:18] NoOp("SIP/4002-0000001d", "TOUCH_MONITOR_OUTPUT=") in new stack
  530. -- Executing [s@macro-hangupcall:19] GotoIf("SIP/4002-0000001d", "1?noautomon2") in new stack
  531. -- Goto (macro-hangupcall,s,25)
  532. -- Executing [s@macro-hangupcall:25] NoOp("SIP/4002-0000001d", "MONITOR_FILENAME=") in new stack
  533. -- Executing [s@macro-hangupcall:26] GotoIf("SIP/4002-0000001d", "1?skiprg") in new stack
  534. -- Goto (macro-hangupcall,s,29)
  535. -- Executing [s@macro-hangupcall:29] GotoIf("SIP/4002-0000001d", "1?skipblkvm") in new stack
  536. -- Goto (macro-hangupcall,s,32)
  537. -- Executing [s@macro-hangupcall:32] GotoIf("SIP/4002-0000001d", "1?theend") in new stack
  538. -- Goto (macro-hangupcall,s,34)
  539. -- Executing [s@macro-hangupcall:34] Hangup("SIP/4002-0000001d", "") in new stack
  540. == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/4002-0000001d' in macro 'hangupcall'
  541. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4002-0000001d'
  542.  
  543. <--- SIP read from UDP:192.168.6.14:5060 --->
  544. ACK sip:94714887904@192.168.6.16 SIP/2.0
  545. Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK-d8754z-73f859fe680c275e-1---d8754z-;rport
  546. Max-Forwards: 70
  547. To: <sip:94714887904@192.168.6.16>;tag=as6c11f5a0
  548. From: "4002"<sip:4002@192.168.6.16>;tag=6b724d3f
  549. Call-ID: OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.
  550. CSeq: 2 ACK
  551. Content-Length: 0
  552.  
  553. <------------->
  554. --- (8 headers 0 lines) ---
  555. Really destroying SIP dialog 'OTExZjUyOWExMWNlNDFjNjhjOWI1YWFhOGIyYjMxZmQ.' Method: ACK
  556. asterisk*CLI> sip set debug off
  557. SIP Debugging Disabled
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