API tools faq
paste
Advertisement
Guest User

Untitled

a guest
Jun 2nd, 2016
170
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 38.03 KB | None | 0 0
raw download clone embed print report
  1. [root@localhost ~]# asterisk -rvvvvvvvvv
  2. Asterisk 13.9.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 13.9.1 currently running on localhost (pid = 1596)
  10. localhost*CLI> sip set debug on
  11. SIP Debugging re-enabled
  12. Reliably Transmitting (no NAT) to 192.168.1.170:5061:
  13. OPTIONS sip:6@192.168.1.170:5061 SIP/2.0
  14. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK088b1b2b
  15. Max-Forwards: 70
  16. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as0b76199c
  17. To: <sip:6@192.168.1.170:5061>
  18. Contact: <sip:Unknown@192.168.1.210:5061>
  19. Call-ID: 1e1a1ec45e20ff3d0ed0b2af05f7bd3c@192.168.1.210:5061
  20. CSeq: 102 OPTIONS
  21. User-Agent: FPBX-13.0.124(13.9.1)
  22. Date: Thu, 02 Jun 2016 20:22:46 GMT
  23. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  24. Supported: replaces, timer
  25. Content-Length: 0
  26.  
  27.  
  28. ---
  29.  
  30. <--- SIP read from UDP:192.168.1.170:5061 --->
  31. SIP/2.0 200 OK
  32. To: <sip:6@192.168.1.170:5061>;tag=817e58dbf025c259i0
  33. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as0b76199c
  34. Call-ID: 1e1a1ec45e20ff3d0ed0b2af05f7bd3c@192.168.1.210:5061
  35. CSeq: 102 OPTIONS
  36. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK088b1b2b
  37. Server: Cisco/SPA501G-7.6.1
  38. Content-Length: 0
  39. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  40. Supported: replaces
  41.  
  42. <------------->
  43. --- (10 headers 0 lines) ---
  44. Really destroying SIP dialog '1e1a1ec45e20ff3d0ed0b2af05f7bd3c@192.168.1.210:5061' Method: OPTIONS
  45. Reliably Transmitting (NAT) to 162.253.134.142:5060:
  46. OPTIONS sip:trunk2.freepbx.com SIP/2.0
  47. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6027557e;rport
  48. Max-Forwards: 70
  49. From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as6002b198
  50. To: <sip:trunk2.freepbx.com>
  51. Contact: <sip:Unknown@71.244.49.87:5061>
  52. Call-ID: 01ac071e52af5bf44ca242076414c119@71.244.49.87:5061
  53. CSeq: 102 OPTIONS
  54. User-Agent: FPBX-13.0.124(13.9.1)
  55. Date: Thu, 02 Jun 2016 20:22:46 GMT
  56. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  57. Supported: replaces, timer
  58. Content-Length: 0
  59.  
  60.  
  61. ---
  62.  
  63. <--- SIP read from UDP:162.253.134.142:5060 --->
  64. SIP/2.0 200 OK
  65. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK6027557e;rport=5061
  66. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as6002b198
  67. To: <sip:trunk2.freepbx.com>;tag=K5B24cQHZjHZp
  68. Call-ID: 01ac071e52af5bf44ca242076414c119@71.244.49.87:5061
  69. CSeq: 102 OPTIONS
  70. Contact: <sip:162.253.134.142>
  71. User-Agent: SIPStation 2.11.3
  72. Accept: application/sdp
  73. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  74. Supported: timer, path, replaces
  75. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  76. Content-Length: 0
  77.  
  78. <------------->
  79. --- (13 headers 0 lines) ---
  80. Really destroying SIP dialog '01ac071e52af5bf44ca242076414c119@71.244.49.87:5061' Method: OPTIONS
  81. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  82. OPTIONS sip:trunk1.freepbx.com SIP/2.0
  83. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK1662c469;rport
  84. Max-Forwards: 70
  85. From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as043deb43
  86. To: <sip:trunk1.freepbx.com>
  87. Contact: <sip:Unknown@71.244.49.87:5061>
  88. Call-ID: 53db861b4cd2345640adc36d47d6d975@71.244.49.87:5061
  89. CSeq: 102 OPTIONS
  90. User-Agent: FPBX-13.0.124(13.9.1)
  91. Date: Thu, 02 Jun 2016 20:22:46 GMT
  92. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  93. Supported: replaces, timer
  94. Content-Length: 0
  95.  
  96.  
  97. ---
  98.  
  99. <--- SIP read from UDP:192.159.66.3:5060 --->
  100. SIP/2.0 200 OK
  101. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK1662c469;rport=5061
  102. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as043deb43
  103. To: <sip:trunk1.freepbx.com>;tag=gyQ1B9K15B9Ka
  104. Call-ID: 53db861b4cd2345640adc36d47d6d975@71.244.49.87:5061
  105. CSeq: 102 OPTIONS
  106. Contact: <sip:192.159.66.3>
  107. User-Agent: SIPStation 2.11.3
  108. Accept: application/sdp
  109. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  110. Supported: timer, path, replaces
  111. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  112. Content-Length: 0
  113.  
  114. <------------->
  115. --- (13 headers 0 lines) ---
  116. Really destroying SIP dialog '53db861b4cd2345640adc36d47d6d975@71.244.49.87:5061' Method: OPTIONS
  117.  
  118. <--- SIP read from UDP:192.168.1.170:5061 --->
  119. INVITE sip:15124610447@192.168.1.210:5061 SIP/2.0
  120. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-eeda14a3
  121. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  122. To: <sip:15124610447@192.168.1.210>
  123. Call-ID: b41cd026-8576898f@192.168.1.170
  124. CSeq: 101 INVITE
  125. Max-Forwards: 70
  126. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  127. Expires: 240
  128. User-Agent: Cisco/SPA501G-7.6.1
  129. Content-Length: 399
  130. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  131. Supported: replaces
  132. Content-Type: application/sdp
  133.  
  134. v=0
  135. o=- 7297278 7297278 IN IP4 192.168.1.170
  136. s=-
  137. c=IN IP4 192.168.1.170
  138. t=0 0
  139. m=audio 16384 RTP/AVP 0 2 8 9 18 96 97 98 101
  140. a=rtpmap:0 PCMU/8000
  141. a=rtpmap:2 G726-32/8000
  142. a=rtpmap:8 PCMA/8000
  143. a=rtpmap:9 G722/8000
  144. a=rtpmap:18 G729a/8000
  145. a=rtpmap:96 G726-40/8000
  146. a=rtpmap:97 G726-24/8000
  147. a=rtpmap:98 G726-16/8000
  148. a=rtpmap:101 telephone-event/8000
  149. a=fmtp:101 0-15
  150. a=ptime:20
  151. a=sendrecv
  152. <------------->
  153. --- (14 headers 18 lines) ---
  154. Sending to 192.168.1.170:5061 (NAT)
  155. Sending to 192.168.1.170:5061 (NAT)
  156. Using INVITE request as basis request - b41cd026-8576898f@192.168.1.170
  157. Found peer '6' for '6' from 192.168.1.170:5061
  158.  
  159. <--- Reliably Transmitting (no NAT) to 192.168.1.170:5061 --->
  160. SIP/2.0 401 Unauthorized
  161. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-eeda14a3;received=192.168.1.170
  162. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  163. To: <sip:15124610447@192.168.1.210>;tag=as74767752
  164. Call-ID: b41cd026-8576898f@192.168.1.170
  165. CSeq: 101 INVITE
  166. Server: FPBX-13.0.124(13.9.1)
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  168. Supported: replaces, timer
  169. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f2d55a7"
  170. Content-Length: 0
  171.  
  172.  
  173. <------------>
  174. Scheduling destruction of SIP dialog 'b41cd026-8576898f@192.168.1.170' in 6400 ms (Method: INVITE)
  175. [2016-06-02 15:22:49] SECURITY[12479]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="ChallengeSent",EventTV="2016-06-02T15:22:49.825-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:6@192.168.1.210",SessionID="0x21c9c88",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",Challenge="4f2d55a7"
  176.  
  177. <--- SIP read from UDP:192.168.1.170:5061 --->
  178. ACK sip:15124610447@192.168.1.210:5061 SIP/2.0
  179. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-eeda14a3
  180. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  181. To: <sip:15124610447@192.168.1.210>;tag=as74767752
  182. Call-ID: b41cd026-8576898f@192.168.1.170
  183. CSeq: 101 ACK
  184. Max-Forwards: 70
  185. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  186. User-Agent: Cisco/SPA501G-7.6.1
  187. Content-Length: 0
  188.  
  189. <------------->
  190. --- (10 headers 0 lines) ---
  191.  
  192. <--- SIP read from UDP:192.168.1.170:5061 --->
  193. INVITE sip:15124610447@192.168.1.210:5061 SIP/2.0
  194. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-1ad25d8f
  195. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  196. To: <sip:15124610447@192.168.1.210>
  197. Call-ID: b41cd026-8576898f@192.168.1.170
  198. CSeq: 102 INVITE
  199. Max-Forwards: 70
  200. Authorization: Digest username="6",realm="asterisk",nonce="4f2d55a7",uri="sip:15124610447@192.168.1.210:5061",algorithm=MD5,response="d0ea0f993da4ca0cd0e2e457eaeb121c"
  201. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  202. Expires: 240
  203. User-Agent: Cisco/SPA501G-7.6.1
  204. Content-Length: 399
  205. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  206. Supported: replaces
  207. Content-Type: application/sdp
  208.  
  209. v=0
  210. o=- 7297278 7297278 IN IP4 192.168.1.170
  211. s=-
  212. c=IN IP4 192.168.1.170
  213. t=0 0
  214. m=audio 16384 RTP/AVP 0 2 8 9 18 96 97 98 101
  215. a=rtpmap:0 PCMU/8000
  216. a=rtpmap:2 G726-32/8000
  217. a=rtpmap:8 PCMA/8000
  218. a=rtpmap:9 G722/8000
  219. a=rtpmap:18 G729a/8000
  220. a=rtpmap:96 G726-40/8000
  221. a=rtpmap:97 G726-24/8000
  222. a=rtpmap:98 G726-16/8000
  223. a=rtpmap:101 telephone-event/8000
  224. a=fmtp:101 0-15
  225. a=ptime:20
  226. a=sendrecv
  227. <------------->
  228. --- (15 headers 18 lines) ---
  229. Sending to 192.168.1.170:5061 (no NAT)
  230. Using INVITE request as basis request - b41cd026-8576898f@192.168.1.170
  231. Found peer '6' for '6' from 192.168.1.170:5061
  232. == Using SIP RTP TOS bits 184
  233. == Using SIP RTP CoS mark 5
  234. Found RTP audio format 0
  235. Found RTP audio format 2
  236. Found RTP audio format 8
  237. Found RTP audio format 9
  238. Found RTP audio format 18
  239. Found RTP audio format 96
  240. Found RTP audio format 97
  241. Found RTP audio format 98
  242. Found RTP audio format 101
  243. Found audio description format PCMU for ID 0
  244. Found audio description format G726-32 for ID 2
  245. Found audio description format PCMA for ID 8
  246. Found audio description format G722 for ID 9
  247. Found audio description format G729a for ID 18
  248. Found unknown media description format G726-40 for ID 96
  249. Found unknown media description format G726-24 for ID 97
  250. Found unknown media description format G726-16 for ID 98
  251. Found audio description format telephone-event for ID 101
  252. Capabilities: us - (ulaw|g722|g729|alaw|speex|opus|g726aal2), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g722|g729|alaw)
  253. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  254. Peer audio RTP is at port 192.168.1.170:16384
  255. Looking for 15124610447 in from-internal (domain 192.168.1.210)
  256. sip_route_dump: route/path hop: <sip:6@192.168.1.170:5061>
  257.  
  258. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  259. SIP/2.0 100 Trying
  260. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-1ad25d8f;received=192.168.1.170
  261. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  262. To: <sip:15124610447@192.168.1.210>
  263. Call-ID: b41cd026-8576898f@192.168.1.170
  264. CSeq: 102 INVITE
  265. Server: FPBX-13.0.124(13.9.1)
  266. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  267. Supported: replaces, timer
  268. Contact: <sip:15124610447@192.168.1.210:5061>
  269. Content-Length: 0
  270.  
  271.  
  272. <------------>
  273. [2016-06-02 15:22:49] SECURITY[12479]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-02T15:22:49.868-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="15124610447",SessionID="0x21c9c88",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",UsingPassword="1"
  274. -- Executing [15124610447@from-internal:1] Macro("SIP/6-00000012", "user-callerid,LIMIT") in new stack
  275. -- Executing [s@macro-user-callerid:1] Set("SIP/6-00000012", "TOUCH_MONITOR=1464898969.18") in new stack
  276. -- Executing [s@macro-user-callerid:2] Set("SIP/6-00000012", "AMPUSER=6") in new stack
  277. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/6-00000012", "0?report") in new stack
  278. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/6-00000012", "1?Set(REALCALLERIDNUM=6)") in new stack
  279. -- Executing [s@macro-user-callerid:5] Set("SIP/6-00000012", "AMPUSER=6") in new stack
  280. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6-00000012", "0?limit") in new stack
  281. -- Executing [s@macro-user-callerid:7] Set("SIP/6-00000012", "AMPUSERCIDNAME=Cisco") in new stack
  282. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/6-00000012", "0?report") in new stack
  283. -- Executing [s@macro-user-callerid:9] Set("SIP/6-00000012", "AMPUSERCID=6") in new stack
  284. -- Executing [s@macro-user-callerid:10] Set("SIP/6-00000012", "__DIAL_OPTIONS=Ttr") in new stack
  285. -- Executing [s@macro-user-callerid:11] Set("SIP/6-00000012", "CALLERID(all)="Cisco" <6>") in new stack
  286. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/6-00000012", "0?limit") in new stack
  287. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/6-00000012", "1?Set(GROUP(concurrency_limit)=6)") in new stack
  288. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/6-00000012", "0?Set(CHANNEL(language)=)") in new stack
  289. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/6-00000012", "1?continue") in new stack
  290. -- Goto (macro-user-callerid,s,29)
  291. -- Executing [s@macro-user-callerid:29] Set("SIP/6-00000012", "CALLERID(number)=6") in new stack
  292. -- Executing [s@macro-user-callerid:30] Set("SIP/6-00000012", "CALLERID(name)=Cisco") in new stack
  293. -- Executing [s@macro-user-callerid:31] Set("SIP/6-00000012", "CDR(cnum)=6") in new stack
  294. -- Executing [s@macro-user-callerid:32] Set("SIP/6-00000012", "CDR(cnam)=Cisco") in new stack
  295. -- Executing [s@macro-user-callerid:33] Set("SIP/6-00000012", "CHANNEL(language)=en") in new stack
  296. -- Executing [15124610447@from-internal:2] Set("SIP/6-00000012", "ROUTEUSER=6") in new stack
  297. -- Executing [15124610447@from-internal:3] GotoIf("SIP/6-00000012", "1?notblind") in new stack
  298. -- Goto (from-internal,15124610447,6)
  299. -- Executing [15124610447@from-internal:6] GotoIf("SIP/6-00000012", "1?restrictedroute-98bd5f7b1447e8791389136169a3a580,15124610447,2:outbound-allroutes,15124610447,2") in new stack
  300. -- Goto (restrictedroute-98bd5f7b1447e8791389136169a3a580,15124610447,2)
  301. -- Executing [15124610447@restrictedroute-98bd5f7b1447e8791389136169a3a580:2] Gosub("SIP/6-00000012", "sub-record-check,s,1(out,15124610447,dontcare)") in new stack
  302. -- Executing [s@sub-record-check:1] GotoIf("SIP/6-00000012", "0?initialized") in new stack
  303. -- Executing [s@sub-record-check:2] Set("SIP/6-00000012", "__REC_STATUS=INITIALIZED") in new stack
  304. -- Executing [s@sub-record-check:3] Set("SIP/6-00000012", "NOW=1464898969") in new stack
  305. -- Executing [s@sub-record-check:4] Set("SIP/6-00000012", "__DAY=02") in new stack
  306. -- Executing [s@sub-record-check:5] Set("SIP/6-00000012", "__MONTH=06") in new stack
  307. -- Executing [s@sub-record-check:6] Set("SIP/6-00000012", "__YEAR=2016") in new stack
  308. -- Executing [s@sub-record-check:7] Set("SIP/6-00000012", "__TIMESTR=20160602-152249") in new stack
  309. -- Executing [s@sub-record-check:8] Set("SIP/6-00000012", "__FROMEXTEN=6") in new stack
  310. -- Executing [s@sub-record-check:9] Set("SIP/6-00000012", "__MON_FMT=wav") in new stack
  311. -- Executing [s@sub-record-check:10] NoOp("SIP/6-00000012", "Recordings initialized") in new stack
  312. -- Executing [s@sub-record-check:11] ExecIf("SIP/6-00000012", "0?Set(ARG3=dontcare)") in new stack
  313. -- Executing [s@sub-record-check:12] Set("SIP/6-00000012", "REC_POLICY_MODE_SAVE=") in new stack
  314. -- Executing [s@sub-record-check:13] ExecIf("SIP/6-00000012", "0?Set(REC_STATUS=NO)") in new stack
  315. -- Executing [s@sub-record-check:14] GotoIf("SIP/6-00000012", "3?checkaction") in new stack
  316. -- Goto (sub-record-check,s,17)
  317. -- Executing [s@sub-record-check:17] GotoIf("SIP/6-00000012", "1?sub-record-check,out,1") in new stack
  318. -- Goto (sub-record-check,out,1)
  319. -- Executing [out@sub-record-check:1] NoOp("SIP/6-00000012", "Outbound Recording Check from 6 to 15124610447") in new stack
  320. -- Executing [out@sub-record-check:2] Set("SIP/6-00000012", "RECMODE=dontcare") in new stack
  321. -- Executing [out@sub-record-check:3] ExecIf("SIP/6-00000012", "1?Goto(routewins)") in new stack
  322. -- Goto (sub-record-check,out,7)
  323. -- Executing [out@sub-record-check:7] Gosub("SIP/6-00000012", "recordcheck,1(dontcare,out,15124610447)") in new stack
  324. -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/6-00000012", "Starting recording check against dontcare") in new stack
  325. -- Executing [recordcheck@sub-record-check:2] Goto("SIP/6-00000012", "dontcare") in new stack
  326. -- Goto (sub-record-check,recordcheck,3)
  327. -- Executing [recordcheck@sub-record-check:3] Return("SIP/6-00000012", "") in new stack
  328. -- Executing [out@sub-record-check:8] Return("SIP/6-00000012", "") in new stack
  329. -- Executing [15124610447@restrictedroute-98bd5f7b1447e8791389136169a3a580:3] ExecIf("SIP/6-00000012", "0 ?Set(CDR(accountcode)=)") in new stack
  330. -- Executing [15124610447@restrictedroute-98bd5f7b1447e8791389136169a3a580:4] Set("SIP/6-00000012", "MOHCLASS=default") in new stack
  331. -- Executing [15124610447@restrictedroute-98bd5f7b1447e8791389136169a3a580:5] Set("SIP/6-00000012", "_NODEST=") in new stack
  332. -- Executing [15124610447@restrictedroute-98bd5f7b1447e8791389136169a3a580:6] Macro("SIP/6-00000012", "dialout-trunk,2,15124610447,,off") in new stack
  333. -- Executing [s@macro-dialout-trunk:1] Set("SIP/6-00000012", "DIAL_TRUNK=2") in new stack
  334. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/6-00000012", "0?sub-pincheck,s,1()") in new stack
  335. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/6-00000012", "0?disabletrunk,1") in new stack
  336. -- Executing [s@macro-dialout-trunk:4] Set("SIP/6-00000012", "DIAL_NUMBER=15124610447") in new stack
  337. -- Executing [s@macro-dialout-trunk:5] Set("SIP/6-00000012", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  338. -- Executing [s@macro-dialout-trunk:6] Set("SIP/6-00000012", "OUTBOUND_GROUP=OUT_2") in new stack
  339. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/6-00000012", "1?nomax") in new stack
  340. -- Goto (macro-dialout-trunk,s,9)
  341. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/6-00000012", "0?skipoutcid") in new stack
  342. -- Executing [s@macro-dialout-trunk:10] Set("SIP/6-00000012", "DIAL_TRUNK_OPTIONS=Tt") in new stack
  343. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/6-00000012", "outbound-callerid,2") in new stack
  344. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/6-00000012", "0?Set(CALLERPRES(name-pres)=)") in new stack
  345. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/6-00000012", "0?Set(CALLERPRES(num-pres)=)") in new stack
  346. -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/6-00000012", "1?Set(REALCALLERIDNUM=6)") in new stack
  347. -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/6-00000012", "1?normcid") in new stack
  348. -- Goto (macro-outbound-callerid,s,7)
  349. -- Executing [s@macro-outbound-callerid:7] Set("SIP/6-00000012", "USEROUTCID=") in new stack
  350. -- Executing [s@macro-outbound-callerid:8] Set("SIP/6-00000012", "EMERGENCYCID=") in new stack
  351. -- Executing [s@macro-outbound-callerid:9] Set("SIP/6-00000012", "TRUNKOUTCID=") in new stack
  352. -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/6-00000012", "1?trunkcid") in new stack
  353. -- Goto (macro-outbound-callerid,s,15)
  354. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/6-00000012", "0?Set(CALLERID(all)=)") in new stack
  355. -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/6-00000012", "0?Set(CALLERID(all)=)") in new stack
  356. -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/6-00000012", "0?Set(CALLERID(all)=)") in new stack
  357. -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/6-00000012", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  358. -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/6-00000012", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  359. -- Executing [s@macro-outbound-callerid:20] Set("SIP/6-00000012", "CDR(outbound_cnum)=6") in new stack
  360. -- Executing [s@macro-outbound-callerid:21] Set("SIP/6-00000012", "CDR(outbound_cnam)=Cisco") in new stack
  361. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/6-00000012", "0?sub-flp-2,s,1()") in new stack
  362. -- Executing [s@macro-dialout-trunk:13] Set("SIP/6-00000012", "OUTNUM=15124610447") in new stack
  363. -- Executing [s@macro-dialout-trunk:14] Set("SIP/6-00000012", "custom=SIP/fpbx-1-cdB7e8PklPds") in new stack
  364. -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6-00000012", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
  365. -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6-00000012", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
  366. -- Executing [s@macro-dialout-trunk:17] Macro("SIP/6-00000012", "dialout-trunk-predial-hook,") in new stack
  367. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6-00000012", "") in new stack
  368. -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6-00000012", "0?bypass,1") in new stack
  369. -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6-00000012", "1?Set(CONNECTEDLINE(num,i)=15124610447)") in new stack
  370. -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6-00000012", "1?Set(CONNECTEDLINE(name,i)=CID:6)") in new stack
  371. -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/6-00000012", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)6)") in new stack
  372. -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/6-00000012", "0?customtrunk") in new stack
  373. -- Executing [s@macro-dialout-trunk:23] Dial("SIP/6-00000012", "SIP/fpbx-1-cdB7e8PklPds/15124610447,300,Tt") in new stack
  374. == Using SIP RTP TOS bits 184
  375. == Using SIP RTP CoS mark 5
  376. Audio is at 10200
  377. Adding codec ulaw to SDP
  378. Adding non-codec 0x1 (telephone-event) to SDP
  379. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  380. INVITE sip:15124610447@trunk1.freepbx.com SIP/2.0
  381. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK2f6964d1;rport
  382. Max-Forwards: 70
  383. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as2cb0f883
  384. To: <sip:15124610447@trunk1.freepbx.com>
  385. Contact: <sip:6@71.244.49.87:5061>
  386. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  387. CSeq: 102 INVITE
  388. User-Agent: FPBX-13.0.124(13.9.1)
  389. Date: Thu, 02 Jun 2016 20:22:49 GMT
  390. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  391. Supported: replaces, timer
  392. Remote-Party-ID: "Cisco" <sip:6@71.244.49.87>;party=calling;privacy=off;screen=no
  393. Content-Type: application/sdp
  394. Content-Length: 249
  395.  
  396. v=0
  397. o=root 499898143 499898143 IN IP4 71.244.49.87
  398. s=Asterisk PBX 13.9.1
  399. c=IN IP4 71.244.49.87
  400. t=0 0
  401. m=audio 10200 RTP/AVP 0 101
  402. a=rtpmap:0 PCMU/8000
  403. a=rtpmap:101 telephone-event/8000
  404. a=fmtp:101 0-16
  405. a=ptime:20
  406. a=maxptime:150
  407. a=sendrecv
  408.  
  409. ---
  410. -- Called SIP/fpbx-1-cdB7e8PklPds/15124610447
  411.  
  412. <--- SIP read from UDP:192.159.66.3:5060 --->
  413. SIP/2.0 100 Trying
  414. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2f6964d1;rport=5061
  415. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  416. To: <sip:15124610447@trunk1.freepbx.com>
  417. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  418. CSeq: 102 INVITE
  419. User-Agent: SIPStation 2.11.3
  420. Content-Length: 0
  421.  
  422. <------------->
  423. --- (8 headers 0 lines) ---
  424.  
  425. <--- SIP read from UDP:192.159.66.3:5060 --->
  426. SIP/2.0 407 Proxy Authentication Required
  427. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2f6964d1;rport=5061
  428. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  429. To: <sip:15124610447@trunk1.freepbx.com>;tag=NHcN9tarc7DQm
  430. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  431. CSeq: 102 INVITE
  432. User-Agent: SIPStation 2.11.3
  433. Accept: application/sdp
  434. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  435. Supported: timer, path, replaces
  436. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  437. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="c6d9afb8-28ff-11e6-bbce-0732f924a662", algorithm=MD5, qop="auth"
  438. Content-Length: 0
  439.  
  440. <------------->
  441. --- (13 headers 0 lines) ---
  442. Transmitting (NAT) to 192.159.66.3:5060:
  443. ACK sip:15124610447@trunk1.freepbx.com SIP/2.0
  444. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK2f6964d1;rport
  445. Max-Forwards: 70
  446. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as2cb0f883
  447. To: <sip:15124610447@trunk1.freepbx.com>;tag=NHcN9tarc7DQm
  448. Contact: <sip:6@71.244.49.87:5061>
  449. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  450. CSeq: 102 ACK
  451. User-Agent: FPBX-13.0.124(13.9.1)
  452. Content-Length: 0
  453.  
  454.  
  455. ---
  456. Audio is at 10200
  457. Adding codec ulaw to SDP
  458. Adding non-codec 0x1 (telephone-event) to SDP
  459. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  460. INVITE sip:15124610447@trunk1.freepbx.com SIP/2.0
  461. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK04f39356;rport
  462. Max-Forwards: 70
  463. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as2cb0f883
  464. To: <sip:15124610447@trunk1.freepbx.com>
  465. Contact: <sip:6@71.244.49.87:5061>
  466. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  467. CSeq: 103 INVITE
  468. User-Agent: FPBX-13.0.124(13.9.1)
  469. Proxy-Authorization: Digest username="cdB7e8PklPds", realm="71.244.49.87", algorithm=MD5, uri="sip:15124610447@trunk1.freepbx.com", nonce="c6d9afb8-28ff-11e6-bbce-0732f924a662", response="f97f7bff242e1a6e5ce6836a70fbc160", qop=auth, cnonce="5b6cd61e", nc=00000001
  470. Date: Thu, 02 Jun 2016 20:22:49 GMT
  471. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  472. Supported: replaces, timer
  473. Remote-Party-ID: "Cisco" <sip:6@71.244.49.87>;party=calling;privacy=off;screen=no
  474. Content-Type: application/sdp
  475. Content-Length: 249
  476.  
  477. v=0
  478. o=root 499898143 499898144 IN IP4 71.244.49.87
  479. s=Asterisk PBX 13.9.1
  480. c=IN IP4 71.244.49.87
  481. t=0 0
  482. m=audio 10200 RTP/AVP 0 101
  483. a=rtpmap:0 PCMU/8000
  484. a=rtpmap:101 telephone-event/8000
  485. a=fmtp:101 0-16
  486. a=ptime:20
  487. a=maxptime:150
  488. a=sendrecv
  489.  
  490. ---
  491.  
  492. <--- SIP read from UDP:192.159.66.3:5060 --->
  493. SIP/2.0 100 Trying
  494. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK04f39356;rport=5061
  495. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  496. To: <sip:15124610447@trunk1.freepbx.com>
  497. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  498. CSeq: 103 INVITE
  499. User-Agent: SIPStation 2.11.3
  500. Content-Length: 0
  501.  
  502. <------------->
  503. --- (8 headers 0 lines) ---
  504.  
  505. <--- SIP read from UDP:192.159.66.3:5060 --->
  506. SIP/2.0 183 Session Progress
  507. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK04f39356;rport=5061
  508. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  509. To: <sip:15124610447@trunk1.freepbx.com>;tag=pt5DBpUU9F49F
  510. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  511. CSeq: 103 INVITE
  512. Contact: <sip:15124610447@192.159.66.3:5060;transport=udp>
  513. User-Agent: SIPStation 2.11.3
  514. Accept: application/sdp
  515. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  516. Supported: timer, path, replaces
  517. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  518. Content-Type: application/sdp
  519. Content-Disposition: session
  520. Content-Length: 223
  521. Remote-Party-ID: "15124610447" <sip:15124610447@trunk1.freepbx.com>;party=calling;privacy=off;screen=no
  522.  
  523. v=0
  524. o=Sonus_UAC 896415 926443 IN IP4 67.231.13.80
  525. s=SIP Media Capabilities
  526. c=IN IP4 67.231.13.80
  527. t=0 0
  528. m=audio 34852 RTP/AVP 0 101
  529. a=rtpmap:0 PCMU/8000
  530. a=rtpmap:101 telephone-event/8000
  531. a=fmtp:101 0-15
  532. a=ptime:20
  533. <------------->
  534. --- (16 headers 10 lines) ---
  535. sip_route_dump: route/path hop: <sip:15124610447@192.159.66.3:5060;transport=udp>
  536. Found RTP audio format 0
  537. Found RTP audio format 101
  538. Found audio description format PCMU for ID 0
  539. Found audio description format telephone-event for ID 101
  540. Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  541. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  542. Peer audio RTP is at port 67.231.13.80:34852
  543. -- SIP/fpbx-1-cdB7e8PklPds-00000013 is making progress passing it to SIP/6-00000012
  544. Audio is at 17308
  545. Adding codec ulaw to SDP
  546. Adding codec g722 to SDP
  547. Adding codec g729 to SDP
  548. Adding codec alaw to SDP
  549. Adding non-codec 0x1 (telephone-event) to SDP
  550.  
  551. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  552. SIP/2.0 183 Session Progress
  553. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-1ad25d8f;received=192.168.1.170
  554. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  555. To: <sip:15124610447@192.168.1.210>;tag=as72c4f51a
  556. Call-ID: b41cd026-8576898f@192.168.1.170
  557. CSeq: 102 INVITE
  558. Server: FPBX-13.0.124(13.9.1)
  559. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  560. Supported: replaces, timer
  561. Contact: <sip:15124610447@192.168.1.210:5061>
  562. Content-Type: application/sdp
  563. Content-Length: 346
  564.  
  565. v=0
  566. o=root 585637713 585637713 IN IP4 192.168.1.210
  567. s=Asterisk PBX 13.9.1
  568. c=IN IP4 192.168.1.210
  569. t=0 0
  570. m=audio 17308 RTP/AVP 0 9 18 8 101
  571. a=rtpmap:0 PCMU/8000
  572. a=rtpmap:9 G722/8000
  573. a=rtpmap:18 G729/8000
  574. a=fmtp:18 annexb=no
  575. a=rtpmap:8 PCMA/8000
  576. a=rtpmap:101 telephone-event/8000
  577. a=fmtp:101 0-16
  578. a=ptime:20
  579. a=maxptime:150
  580. a=sendrecv
  581.  
  582. <------------>
  583. > 0x21e73b0 -- Probation passed - setting RTP source address to 192.168.1.170:16384
  584.  
  585. <--- SIP read from UDP:192.168.1.170:5061 --->
  586. NOTIFY sip:192.168.1.210:5061 SIP/2.0
  587. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-4a57b5fa
  588. From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
  589. To: <sip:192.168.1.210>
  590. Call-ID: 5f2a7c5c-9da78ccd@192.168.1.170
  591. CSeq: 4860 NOTIFY
  592. Max-Forwards: 70
  593. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  594. Event: keep-alive
  595. User-Agent: Cisco/SPA501G-7.6.1
  596. Content-Length: 0
  597.  
  598. <------------->
  599. --- (11 headers 0 lines) ---
  600.  
  601. <--- Transmitting (NAT) to 192.168.1.170:5061 --->
  602. SIP/2.0 200 OK
  603. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-4a57b5fa;received=192.168.1.170;rport=5061
  604. From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
  605. To: <sip:192.168.1.210>;tag=as29f6e0d7
  606. Call-ID: 5f2a7c5c-9da78ccd@192.168.1.170
  607. CSeq: 4860 NOTIFY
  608. Server: FPBX-13.0.124(13.9.1)
  609. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  610. Supported: replaces, timer
  611. Content-Length: 0
  612.  
  613.  
  614. <------------>
  615. Scheduling destruction of SIP dialog '5f2a7c5c-9da78ccd@192.168.1.170' in 32000 ms (Method: NOTIFY)
  616.  
  617. <--- SIP read from UDP:192.159.66.3:5060 --->
  618. SIP/2.0 200 OK
  619. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK04f39356;rport=5061
  620. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  621. To: <sip:15124610447@trunk1.freepbx.com>;tag=pt5DBpUU9F49F
  622. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  623. CSeq: 103 INVITE
  624. Contact: <sip:15124610447@192.159.66.3:5060;transport=udp>
  625. User-Agent: SIPStation 2.11.3
  626. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  627. Supported: timer, path, replaces
  628. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  629. Content-Type: application/sdp
  630. Content-Disposition: session
  631. Content-Length: 223
  632. Remote-Party-ID: "Outbound Call" <sip:+15124610447@trunk1.freepbx.com>;party=calling;privacy=off;screen=no
  633.  
  634. v=0
  635. o=Sonus_UAC 896415 926443 IN IP4 67.231.13.80
  636. s=SIP Media Capabilities
  637. c=IN IP4 67.231.13.80
  638. t=0 0
  639. m=audio 34852 RTP/AVP 0 101
  640. a=rtpmap:0 PCMU/8000
  641. a=rtpmap:101 telephone-event/8000
  642. a=fmtp:101 0-15
  643. a=ptime:20
  644. <------------->
  645. --- (15 headers 10 lines) ---
  646. sip_route_dump: route/path hop: <sip:15124610447@192.159.66.3:5060;transport=udp>
  647. Transmitting (NAT) to 192.159.66.3:5060:
  648. ACK sip:15124610447@192.159.66.3:5060;transport=udp SIP/2.0
  649. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK4af63c53;rport
  650. Max-Forwards: 70
  651. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as2cb0f883
  652. To: <sip:15124610447@trunk1.freepbx.com>;tag=pt5DBpUU9F49F
  653. Contact: <sip:6@71.244.49.87:5061>
  654. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  655. CSeq: 103 ACK
  656. User-Agent: FPBX-13.0.124(13.9.1)
  657. Content-Length: 0
  658.  
  659.  
  660. ---
  661. -- SIP/fpbx-1-cdB7e8PklPds-00000013 answered SIP/6-00000012
  662. Audio is at 17308
  663. Adding codec ulaw to SDP
  664. Adding codec g722 to SDP
  665. Adding codec g729 to SDP
  666. Adding codec alaw to SDP
  667. Adding non-codec 0x1 (telephone-event) to SDP
  668.  
  669. <--- Reliably Transmitting (no NAT) to 192.168.1.170:5061 --->
  670. SIP/2.0 200 OK
  671. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-1ad25d8f;received=192.168.1.170
  672. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  673. To: <sip:15124610447@192.168.1.210>;tag=as72c4f51a
  674. Call-ID: b41cd026-8576898f@192.168.1.170
  675. CSeq: 102 INVITE
  676. Server: FPBX-13.0.124(13.9.1)
  677. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  678. Supported: replaces, timer
  679. Contact: <sip:15124610447@192.168.1.210:5061>
  680. Content-Type: application/sdp
  681. Content-Length: 346
  682.  
  683. v=0
  684. o=root 585637713 585637713 IN IP4 192.168.1.210
  685. s=Asterisk PBX 13.9.1
  686. c=IN IP4 192.168.1.210
  687. t=0 0
  688. m=audio 17308 RTP/AVP 0 9 18 8 101
  689. a=rtpmap:0 PCMU/8000
  690. a=rtpmap:9 G722/8000
  691. a=rtpmap:18 G729/8000
  692. a=fmtp:18 annexb=no
  693. a=rtpmap:8 PCMA/8000
  694. a=rtpmap:101 telephone-event/8000
  695. a=fmtp:101 0-16
  696. a=ptime:20
  697. a=maxptime:150
  698. a=sendrecv
  699.  
  700. <------------>
  701. -- Channel SIP/fpbx-1-cdB7e8PklPds-00000013 joined 'simple_bridge' basic-bridge <f12edc0c-feb0-43bf-b917-b9766aec9ee6>
  702. -- Channel SIP/6-00000012 joined 'simple_bridge' basic-bridge <f12edc0c-feb0-43bf-b917-b9766aec9ee6>
  703.  
  704. <--- SIP read from UDP:192.168.1.170:5061 --->
  705. ACK sip:15124610447@192.168.1.210:5061 SIP/2.0
  706. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-dd8ac0ca
  707. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  708. To: <sip:15124610447@192.168.1.210>;tag=as72c4f51a
  709. Call-ID: b41cd026-8576898f@192.168.1.170
  710. CSeq: 102 ACK
  711. Max-Forwards: 70
  712. Authorization: Digest username="6",realm="asterisk",nonce="4f2d55a7",uri="sip:15124610447@192.168.1.210:5061",algorithm=MD5,response="d0ea0f993da4ca0cd0e2e457eaeb121c"
  713. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  714. User-Agent: Cisco/SPA501G-7.6.1
  715. Content-Length: 0
  716.  
  717. <------------->
  718. --- (11 headers 0 lines) ---
  719.  
  720. <--- SIP read from UDP:192.168.1.170:5061 --->
  721. BYE sip:15124610447@192.168.1.210:5061 SIP/2.0
  722. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-5b4eb50a
  723. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  724. To: <sip:15124610447@192.168.1.210>;tag=as72c4f51a
  725. Call-ID: b41cd026-8576898f@192.168.1.170
  726. CSeq: 103 BYE
  727. Max-Forwards: 70
  728. Authorization: Digest username="6",realm="asterisk",nonce="4f2d55a7",uri="sip:15124610447@192.168.1.210:5061",algorithm=MD5,response="1fb55ff9881ae0c2e6f38aec0732afa9"
  729. User-Agent: Cisco/SPA501G-7.6.1
  730. Content-Length: 0
  731.  
  732. <------------->
  733. --- (10 headers 0 lines) ---
  734. Sending to 192.168.1.170:5061 (no NAT)
  735. Scheduling destruction of SIP dialog 'b41cd026-8576898f@192.168.1.170' in 6400 ms (Method: BYE)
  736.  
  737. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  738. SIP/2.0 200 OK
  739. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-5b4eb50a;received=192.168.1.170
  740. From: "Cisco" <sip:6@192.168.1.210>;tag=be21773a9cc9ed8bo0
  741. To: <sip:15124610447@192.168.1.210>;tag=as72c4f51a
  742. Call-ID: b41cd026-8576898f@192.168.1.170
  743. CSeq: 103 BYE
  744. Server: FPBX-13.0.124(13.9.1)
  745. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  746. Supported: replaces, timer
  747. Content-Length: 0
  748.  
  749.  
  750. <------------>
  751. -- Channel SIP/6-00000012 left 'simple_bridge' basic-bridge <f12edc0c-feb0-43bf-b917-b9766aec9ee6>
  752. -- Channel SIP/fpbx-1-cdB7e8PklPds-00000013 left 'simple_bridge' basic-bridge <f12edc0c-feb0-43bf-b917-b9766aec9ee6>
  753. Scheduling destruction of SIP dialog '0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061' in 6400 ms (Method: INVITE)
  754. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  755. BYE sip:15124610447@192.159.66.3:5060;transport=udp SIP/2.0
  756. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK2a8a4af9;rport
  757. Max-Forwards: 70
  758. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as2cb0f883
  759. To: <sip:15124610447@trunk1.freepbx.com>;tag=pt5DBpUU9F49F
  760. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  761. CSeq: 104 BYE
  762. User-Agent: FPBX-13.0.124(13.9.1)
  763. Proxy-Authorization: Digest username="cdB7e8PklPds", realm="71.244.49.87", algorithm=MD5, uri="sip:15124610447@192.159.66.3:5060", nonce="c6d9afb8-28ff-11e6-bbce-0732f924a662", response="f634e730533dd5532ab9097bd99eb1c4", qop=auth, cnonce="494fab9e", nc=00000002
  764. X-Asterisk-HangupCause: Normal Clearing
  765. X-Asterisk-HangupCauseCode: 16
  766. Content-Length: 0
  767.  
  768.  
  769. ---
  770. == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/6-00000012' in macro 'dialout-trunk'
  771. == Spawn extension (restrictedroute-98bd5f7b1447e8791389136169a3a580, 15124610447, 6) exited non-zero on 'SIP/6-00000012'
  772. -- Executing [h@restrictedroute-98bd5f7b1447e8791389136169a3a580:1] Hangup("SIP/6-00000012", "") in new stack
  773. == Spawn extension (restrictedroute-98bd5f7b1447e8791389136169a3a580, h, 1) exited non-zero on 'SIP/6-00000012'
  774.  
  775. <--- SIP read from UDP:192.159.66.3:5060 --->
  776. SIP/2.0 200 OK
  777. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2a8a4af9;rport=5061
  778. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as2cb0f883
  779. To: <sip:15124610447@trunk1.freepbx.com>;tag=pt5DBpUU9F49F
  780. Call-ID: 0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061
  781. CSeq: 104 BYE
  782. User-Agent: SIPStation 2.11.3
  783. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  784. Supported: timer, path, replaces
  785. Content-Length: 0
  786.  
  787. <------------->
  788. --- (10 headers 0 lines) ---
  789. Really destroying SIP dialog '0ca5c85653efe0583c1ef0b1000eabe0@71.244.49.87:5061' Method: INVITE
  790. [2016-06-02 15:23:01] SECURITY[12479]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-02T15:23:01.568-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2314e28",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/33729",UsingPassword="0",SessionTV="2016-06-02T15:23:01.568-0500"
  791. [2016-06-02 15:23:01] SECURITY[12479]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-02T15:23:01.851-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2314e28",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/33731",UsingPassword="0",SessionTV="2016-06-02T15:23:01.851-0500"
  792. [2016-06-02 15:23:01] SECURITY[12479]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-02T15:23:01.854-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x1e471c8",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/33733",UsingPassword="0",SessionTV="2016-06-02T15:23:01.854-0500"
  793. localhost*CLI>
  794. Disconnected from Asterisk server
  795. Asterisk cleanly ending (0).
  796. Executing last minute cleanups
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!