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  1. [root@localhost ~]# asterisk -rvvvvvvvvv
  2. Asterisk 13.7.1, 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 13.7.1 currently running on localhost (pid = 1596)
  10. localhost*CLI> sip set debug on
  11. SIP Debugging enabled
  12.  
  13. <--- SIP read from UDP:192.168.1.170:5061 --->
  14. NOTIFY sip:192.168.1.210:5061 SIP/2.0
  15. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-369e7c58
  16. From: "Cisco" <sip:6@192.168.1.210>;tag=4e3052ad8cb6c2a8o0
  17. To: <sip:192.168.1.210>
  18. Call-ID: ec088d91-edff2514@192.168.1.170
  19. CSeq: 35976 NOTIFY
  20. Max-Forwards: 70
  21. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  22. Event: keep-alive
  23. User-Agent: Cisco/SPA501G-7.6.1
  24. Content-Length: 0
  25.  
  26. <------------->
  27. --- (11 headers 0 lines) ---
  28.  
  29. <--- Transmitting (NAT) to 192.168.1.170:5061 --->
  30. SIP/2.0 200 OK
  31. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-369e7c58;received=192.168.1.170;rport=5061
  32. From: "Cisco" <sip:6@192.168.1.210>;tag=4e3052ad8cb6c2a8o0
  33. To: <sip:192.168.1.210>;tag=as3d1464ff
  34. Call-ID: ec088d91-edff2514@192.168.1.170
  35. CSeq: 35976 NOTIFY
  36. Server: FPBX-13.0.120(13.7.1)
  37. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  38. Supported: replaces, timer
  39. Content-Length: 0
  40.  
  41.  
  42. <------------>
  43. Scheduling destruction of SIP dialog 'ec088d91-edff2514@192.168.1.170' in 32000 ms (Method: NOTIFY)
  44. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  45. OPTIONS sip:trunk1.freepbx.com SIP/2.0
  46. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6d63587c;rport
  47. Max-Forwards: 70
  48. From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as72032221
  49. To: <sip:trunk1.freepbx.com>
  50. Contact: <sip:Unknown@71.244.49.87:5061>
  51. Call-ID: 582935797b1ca47639646af47cfaf5e7@71.244.49.87:5061
  52. CSeq: 102 OPTIONS
  53. User-Agent: FPBX-13.0.120(13.7.1)
  54. Date: Wed, 01 Jun 2016 22:19:58 GMT
  55. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  56. Supported: replaces, timer
  57. Content-Length: 0
  58.  
  59.  
  60. ---
  61.  
  62. <--- SIP read from UDP:192.159.66.3:5060 --->
  63. SIP/2.0 200 OK
  64. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK6d63587c;rport=5061
  65. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as72032221
  66. To: <sip:trunk1.freepbx.com>;tag=8gyeFQ0ye3t3j
  67. Call-ID: 582935797b1ca47639646af47cfaf5e7@71.244.49.87:5061
  68. CSeq: 102 OPTIONS
  69. Contact: <sip:192.159.66.3>
  70. User-Agent: SIPStation 2.11.3
  71. Accept: application/sdp
  72. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  73. Supported: timer, path, replaces
  74. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  75. Content-Length: 0
  76.  
  77. <------------->
  78. --- (13 headers 0 lines) ---
  79. Really destroying SIP dialog '582935797b1ca47639646af47cfaf5e7@71.244.49.87:5061' Method: OPTIONS
  80. [2016-06-01 17:19:59] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:19:59.827-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49931",UsingPassword="0",SessionTV="2016-06-01T17:19:59.827-0500"
  81. [2016-06-01 17:20:01] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:01.358-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49933",UsingPassword="0",SessionTV="2016-06-01T17:20:01.358-0500"
  82. [2016-06-01 17:20:01] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:01.362-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x312c818",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49936",UsingPassword="0",SessionTV="2016-06-01T17:20:01.362-0500"
  83. [2016-06-01 17:20:01] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:01.365-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e53ac8",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49937",UsingPassword="0",SessionTV="2016-06-01T17:20:01.365-0500"
  84. [2016-06-01 17:20:01] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:01.370-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2f00568",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49939",UsingPassword="0",SessionTV="2016-06-01T17:20:01.370-0500"
  85. -- Remote UNIX connection
  86. -- Remote UNIX connection disconnected
  87. -- Remote UNIX connection
  88. -- Remote UNIX connection disconnected
  89. [2016-06-01 17:20:03] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:03.480-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49941",UsingPassword="0",SessionTV="2016-06-01T17:20:03.480-0500"
  90. [2016-06-01 17:20:04] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:04.832-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49943",UsingPassword="0",SessionTV="2016-06-01T17:20:04.832-0500"
  91.  
  92. <--- SIP read from UDP:192.168.1.170:5061 --->
  93. INVITE sip:5124614444@192.168.1.210:5061 SIP/2.0
  94. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-68195358
  95. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  96. To: <sip:5124614444@192.168.1.210>
  97. Call-ID: 8128a648-945b0518@192.168.1.170
  98. CSeq: 101 INVITE
  99. Max-Forwards: 70
  100. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  101. Expires: 240
  102. User-Agent: Cisco/SPA501G-7.6.1
  103. Content-Length: 401
  104. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  105. Supported: replaces
  106. Content-Type: application/sdp
  107.  
  108. v=0
  109. o=- 51610689 51610689 IN IP4 192.168.1.170
  110. s=-
  111. c=IN IP4 192.168.1.170
  112. t=0 0
  113. m=audio 16440 RTP/AVP 0 2 8 9 18 96 97 98 101
  114. a=rtpmap:0 PCMU/8000
  115. a=rtpmap:2 G726-32/8000
  116. a=rtpmap:8 PCMA/8000
  117. a=rtpmap:9 G722/8000
  118. a=rtpmap:18 G729a/8000
  119. a=rtpmap:96 G726-40/8000
  120. a=rtpmap:97 G726-24/8000
  121. a=rtpmap:98 G726-16/8000
  122. a=rtpmap:101 telephone-event/8000
  123. a=fmtp:101 0-15
  124. a=ptime:20
  125. a=sendrecv
  126. <------------->
  127. --- (14 headers 18 lines) ---
  128. Sending to 192.168.1.170:5061 (NAT)
  129. Sending to 192.168.1.170:5061 (NAT)
  130. Using INVITE request as basis request - 8128a648-945b0518@192.168.1.170
  131. Found peer '6' for '6' from 192.168.1.170:5061
  132.  
  133. <--- Reliably Transmitting (no NAT) to 192.168.1.170:5061 --->
  134. SIP/2.0 401 Unauthorized
  135. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-68195358;received=192.168.1.170
  136. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  137. To: <sip:5124614444@192.168.1.210>;tag=as377b9c1d
  138. Call-ID: 8128a648-945b0518@192.168.1.170
  139. CSeq: 101 INVITE
  140. Server: FPBX-13.0.120(13.7.1)
  141. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  142. Supported: replaces, timer
  143. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23998573"
  144. Content-Length: 0
  145.  
  146.  
  147. <------------>
  148. Scheduling destruction of SIP dialog '8128a648-945b0518@192.168.1.170' in 6400 ms (Method: INVITE)
  149. [2016-06-01 17:20:08] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="ChallengeSent",EventTV="2016-06-01T17:20:08.720-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:6@192.168.1.210",SessionID="0x2f93738",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",Challenge="23998573"
  150.  
  151. <--- SIP read from UDP:192.168.1.170:5061 --->
  152. ACK sip:5124614444@192.168.1.210:5061 SIP/2.0
  153. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-68195358
  154. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  155. To: <sip:5124614444@192.168.1.210>;tag=as377b9c1d
  156. Call-ID: 8128a648-945b0518@192.168.1.170
  157. CSeq: 101 ACK
  158. Max-Forwards: 70
  159. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  160. User-Agent: Cisco/SPA501G-7.6.1
  161. Content-Length: 0
  162.  
  163. <------------->
  164. --- (10 headers 0 lines) ---
  165.  
  166. <--- SIP read from UDP:192.168.1.170:5061 --->
  167. INVITE sip:5124614444@192.168.1.210:5061 SIP/2.0
  168. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-c41153e8
  169. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  170. To: <sip:5124614444@192.168.1.210>
  171. Call-ID: 8128a648-945b0518@192.168.1.170
  172. CSeq: 102 INVITE
  173. Max-Forwards: 70
  174. Authorization: Digest username="6",realm="asterisk",nonce="23998573",uri="sip:5124614444@192.168.1.210:5061",algorithm=MD5,response="13b76bdbf26951c3a4a55a42458975ab"
  175. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  176. Expires: 240
  177. User-Agent: Cisco/SPA501G-7.6.1
  178. Content-Length: 401
  179. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  180. Supported: replaces
  181. Content-Type: application/sdp
  182.  
  183. v=0
  184. o=- 51610689 51610689 IN IP4 192.168.1.170
  185. s=-
  186. c=IN IP4 192.168.1.170
  187. t=0 0
  188. m=audio 16440 RTP/AVP 0 2 8 9 18 96 97 98 101
  189. a=rtpmap:0 PCMU/8000
  190. a=rtpmap:2 G726-32/8000
  191. a=rtpmap:8 PCMA/8000
  192. a=rtpmap:9 G722/8000
  193. a=rtpmap:18 G729a/8000
  194. a=rtpmap:96 G726-40/8000
  195. a=rtpmap:97 G726-24/8000
  196. a=rtpmap:98 G726-16/8000
  197. a=rtpmap:101 telephone-event/8000
  198. a=fmtp:101 0-15
  199. a=ptime:20
  200. a=sendrecv
  201. <------------->
  202. --- (15 headers 18 lines) ---
  203. Sending to 192.168.1.170:5061 (no NAT)
  204. Using INVITE request as basis request - 8128a648-945b0518@192.168.1.170
  205. Found peer '6' for '6' from 192.168.1.170:5061
  206. == Using SIP RTP TOS bits 184
  207. == Using SIP RTP CoS mark 5
  208. Found RTP audio format 0
  209. Found RTP audio format 2
  210. Found RTP audio format 8
  211. Found RTP audio format 9
  212. Found RTP audio format 18
  213. Found RTP audio format 96
  214. Found RTP audio format 97
  215. Found RTP audio format 98
  216. Found RTP audio format 101
  217. Found audio description format PCMU for ID 0
  218. Found audio description format G726-32 for ID 2
  219. Found audio description format PCMA for ID 8
  220. Found audio description format G722 for ID 9
  221. Found audio description format G729a for ID 18
  222. Found unknown media description format G726-40 for ID 96
  223. Found unknown media description format G726-24 for ID 97
  224. Found unknown media description format G726-16 for ID 98
  225. Found audio description format telephone-event for ID 101
  226. 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)
  227. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  228. Peer audio RTP is at port 192.168.1.170:16440
  229. Looking for 5124614444 in from-internal (domain 192.168.1.210)
  230. sip_route_dump: route/path hop: <sip:6@192.168.1.170:5061>
  231.  
  232. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  233. SIP/2.0 100 Trying
  234. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-c41153e8;received=192.168.1.170
  235. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  236. To: <sip:5124614444@192.168.1.210>
  237. Call-ID: 8128a648-945b0518@192.168.1.170
  238. CSeq: 102 INVITE
  239. Server: FPBX-13.0.120(13.7.1)
  240. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  241. Supported: replaces, timer
  242. Contact: <sip:5124614444@192.168.1.210:5061>
  243. Content-Length: 0
  244.  
  245.  
  246. <------------>
  247. [2016-06-01 17:20:08] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:08.763-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="5124614444",SessionID="0x2f93738",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",UsingPassword="1"
  248. -- Executing [5124614444@from-internal:1] Macro("SIP/6-0000001c", "user-callerid,LIMIT") in new stack
  249. -- Executing [s@macro-user-callerid:1] Set("SIP/6-0000001c", "TOUCH_MONITOR=1464819608.28") in new stack
  250. -- Executing [s@macro-user-callerid:2] Set("SIP/6-0000001c", "AMPUSER=6") in new stack
  251. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/6-0000001c", "0?report") in new stack
  252. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/6-0000001c", "1?Set(REALCALLERIDNUM=6)") in new stack
  253. -- Executing [s@macro-user-callerid:5] Set("SIP/6-0000001c", "AMPUSER=6") in new stack
  254. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6-0000001c", "0?limit") in new stack
  255. -- Executing [s@macro-user-callerid:7] Set("SIP/6-0000001c", "AMPUSERCIDNAME=Cisco") in new stack
  256. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/6-0000001c", "0?report") in new stack
  257. -- Executing [s@macro-user-callerid:9] Set("SIP/6-0000001c", "AMPUSERCID=6") in new stack
  258. -- Executing [s@macro-user-callerid:10] Set("SIP/6-0000001c", "__DIAL_OPTIONS=Ttr") in new stack
  259. -- Executing [s@macro-user-callerid:11] Set("SIP/6-0000001c", "CALLERID(all)="Cisco" <6>") in new stack
  260. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/6-0000001c", "0?limit") in new stack
  261. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/6-0000001c", "1?Set(GROUP(concurrency_limit)=6)") in new stack
  262. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/6-0000001c", "0?Set(CHANNEL(language)=)") in new stack
  263. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/6-0000001c", "1?continue") in new stack
  264. -- Goto (macro-user-callerid,s,29)
  265. -- Executing [s@macro-user-callerid:29] Set("SIP/6-0000001c", "CALLERID(number)=6") in new stack
  266. -- Executing [s@macro-user-callerid:30] Set("SIP/6-0000001c", "CALLERID(name)=Cisco") in new stack
  267. -- Executing [s@macro-user-callerid:31] Set("SIP/6-0000001c", "CDR(cnum)=6") in new stack
  268. -- Executing [s@macro-user-callerid:32] Set("SIP/6-0000001c", "CDR(cnam)=Cisco") in new stack
  269. -- Executing [s@macro-user-callerid:33] Set("SIP/6-0000001c", "CHANNEL(language)=en") in new stack
  270. -- Executing [5124614444@from-internal:2] Set("SIP/6-0000001c", "ROUTEUSER=6") in new stack
  271. -- Executing [5124614444@from-internal:3] GotoIf("SIP/6-0000001c", "1?notblind") in new stack
  272. -- Goto (from-internal,5124614444,6)
  273. -- Executing [5124614444@from-internal:6] GotoIf("SIP/6-0000001c", "1?restrictedroute-98bd5f7b1447e8791389136169a3a580,5124614444,2:outbound-allroutes,5124614444,2") in new stack
  274. -- Goto (restrictedroute-98bd5f7b1447e8791389136169a3a580,5124614444,2)
  275. -- Executing [5124614444@restrictedroute-98bd5f7b1447e8791389136169a3a580:2] Gosub("SIP/6-0000001c", "sub-record-check,s,1(out,5124614444,dontcare)") in new stack
  276. -- Executing [s@sub-record-check:1] GotoIf("SIP/6-0000001c", "0?initialized") in new stack
  277. -- Executing [s@sub-record-check:2] Set("SIP/6-0000001c", "__REC_STATUS=INITIALIZED") in new stack
  278. -- Executing [s@sub-record-check:3] Set("SIP/6-0000001c", "NOW=1464819608") in new stack
  279. -- Executing [s@sub-record-check:4] Set("SIP/6-0000001c", "__DAY=01") in new stack
  280. -- Executing [s@sub-record-check:5] Set("SIP/6-0000001c", "__MONTH=06") in new stack
  281. -- Executing [s@sub-record-check:6] Set("SIP/6-0000001c", "__YEAR=2016") in new stack
  282. -- Executing [s@sub-record-check:7] Set("SIP/6-0000001c", "__TIMESTR=20160601-172008") in new stack
  283. -- Executing [s@sub-record-check:8] Set("SIP/6-0000001c", "__FROMEXTEN=6") in new stack
  284. -- Executing [s@sub-record-check:9] Set("SIP/6-0000001c", "__MON_FMT=wav") in new stack
  285. -- Executing [s@sub-record-check:10] NoOp("SIP/6-0000001c", "Recordings initialized") in new stack
  286. -- Executing [s@sub-record-check:11] ExecIf("SIP/6-0000001c", "0?Set(ARG3=dontcare)") in new stack
  287. -- Executing [s@sub-record-check:12] Set("SIP/6-0000001c", "REC_POLICY_MODE_SAVE=") in new stack
  288. -- Executing [s@sub-record-check:13] ExecIf("SIP/6-0000001c", "0?Set(REC_STATUS=NO)") in new stack
  289. -- Executing [s@sub-record-check:14] GotoIf("SIP/6-0000001c", "3?checkaction") in new stack
  290. -- Goto (sub-record-check,s,17)
  291. -- Executing [s@sub-record-check:17] GotoIf("SIP/6-0000001c", "1?sub-record-check,out,1") in new stack
  292. -- Goto (sub-record-check,out,1)
  293. -- Executing [out@sub-record-check:1] NoOp("SIP/6-0000001c", "Outbound Recording Check from 6 to 5124614444") in new stack
  294. -- Executing [out@sub-record-check:2] Set("SIP/6-0000001c", "RECMODE=dontcare") in new stack
  295. -- Executing [out@sub-record-check:3] ExecIf("SIP/6-0000001c", "1?Goto(routewins)") in new stack
  296. -- Goto (sub-record-check,out,7)
  297. -- Executing [out@sub-record-check:7] Gosub("SIP/6-0000001c", "recordcheck,1(dontcare,out,5124614444)") in new stack
  298. -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/6-0000001c", "Starting recording check against dontcare") in new stack
  299. -- Executing [recordcheck@sub-record-check:2] Goto("SIP/6-0000001c", "dontcare") in new stack
  300. -- Goto (sub-record-check,recordcheck,3)
  301. -- Executing [recordcheck@sub-record-check:3] Return("SIP/6-0000001c", "") in new stack
  302. -- Executing [out@sub-record-check:8] Return("SIP/6-0000001c", "") in new stack
  303. -- Executing [5124614444@restrictedroute-98bd5f7b1447e8791389136169a3a580:3] ExecIf("SIP/6-0000001c", "0 ?Set(CDR(accountcode)=)") in new stack
  304. -- Executing [5124614444@restrictedroute-98bd5f7b1447e8791389136169a3a580:4] Set("SIP/6-0000001c", "MOHCLASS=default") in new stack
  305. -- Executing [5124614444@restrictedroute-98bd5f7b1447e8791389136169a3a580:5] Set("SIP/6-0000001c", "_NODEST=") in new stack
  306. -- Executing [5124614444@restrictedroute-98bd5f7b1447e8791389136169a3a580:6] Macro("SIP/6-0000001c", "dialout-trunk,2,5124614444,,off") in new stack
  307. -- Executing [s@macro-dialout-trunk:1] Set("SIP/6-0000001c", "DIAL_TRUNK=2") in new stack
  308. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/6-0000001c", "0?sub-pincheck,s,1()") in new stack
  309. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/6-0000001c", "0?disabletrunk,1") in new stack
  310. -- Executing [s@macro-dialout-trunk:4] Set("SIP/6-0000001c", "DIAL_NUMBER=5124614444") in new stack
  311. -- Executing [s@macro-dialout-trunk:5] Set("SIP/6-0000001c", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  312. -- Executing [s@macro-dialout-trunk:6] Set("SIP/6-0000001c", "OUTBOUND_GROUP=OUT_2") in new stack
  313. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/6-0000001c", "1?nomax") in new stack
  314. -- Goto (macro-dialout-trunk,s,9)
  315. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/6-0000001c", "0?skipoutcid") in new stack
  316. -- Executing [s@macro-dialout-trunk:10] Set("SIP/6-0000001c", "DIAL_TRUNK_OPTIONS=Tt") in new stack
  317. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/6-0000001c", "outbound-callerid,2") in new stack
  318. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/6-0000001c", "0?Set(CALLERPRES(name-pres)=)") in new stack
  319. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/6-0000001c", "0?Set(CALLERPRES(num-pres)=)") in new stack
  320. -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/6-0000001c", "1?Set(REALCALLERIDNUM=6)") in new stack
  321. -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/6-0000001c", "1?normcid") in new stack
  322. -- Goto (macro-outbound-callerid,s,7)
  323. -- Executing [s@macro-outbound-callerid:7] Set("SIP/6-0000001c", "USEROUTCID=") in new stack
  324. -- Executing [s@macro-outbound-callerid:8] Set("SIP/6-0000001c", "EMERGENCYCID=") in new stack
  325. -- Executing [s@macro-outbound-callerid:9] Set("SIP/6-0000001c", "TRUNKOUTCID=") in new stack
  326. -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/6-0000001c", "1?trunkcid") in new stack
  327. -- Goto (macro-outbound-callerid,s,15)
  328. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/6-0000001c", "0?Set(CALLERID(all)=)") in new stack
  329. -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/6-0000001c", "0?Set(CALLERID(all)=)") in new stack
  330. -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/6-0000001c", "0?Set(CALLERID(all)=)") in new stack
  331. -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/6-0000001c", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  332. -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/6-0000001c", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  333. -- Executing [s@macro-outbound-callerid:20] Set("SIP/6-0000001c", "CDR(outbound_cnum)=6") in new stack
  334. -- Executing [s@macro-outbound-callerid:21] Set("SIP/6-0000001c", "CDR(outbound_cnam)=Cisco") in new stack
  335. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/6-0000001c", "0?sub-flp-2,s,1()") in new stack
  336. -- Executing [s@macro-dialout-trunk:13] Set("SIP/6-0000001c", "OUTNUM=5124614444") in new stack
  337. -- Executing [s@macro-dialout-trunk:14] Set("SIP/6-0000001c", "custom=SIP/fpbx-1-cdB7e8PklPds") in new stack
  338. -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6-0000001c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
  339. -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6-0000001c", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
  340. -- Executing [s@macro-dialout-trunk:17] Macro("SIP/6-0000001c", "dialout-trunk-predial-hook,") in new stack
  341. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6-0000001c", "") in new stack
  342. -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6-0000001c", "0?bypass,1") in new stack
  343. -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6-0000001c", "1?Set(CONNECTEDLINE(num,i)=5124614444)") in new stack
  344. -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6-0000001c", "1?Set(CONNECTEDLINE(name,i)=CID:6)") in new stack
  345. -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/6-0000001c", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)6)") in new stack
  346. -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/6-0000001c", "0?customtrunk") in new stack
  347. -- Executing [s@macro-dialout-trunk:23] Dial("SIP/6-0000001c", "SIP/fpbx-1-cdB7e8PklPds/5124614444,300,Tt") in new stack
  348. == Using SIP RTP TOS bits 184
  349. == Using SIP RTP CoS mark 5
  350. Audio is at 13292
  351. Adding codec ulaw to SDP
  352. Adding non-codec 0x1 (telephone-event) to SDP
  353. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  354. INVITE sip:5124614444@trunk1.freepbx.com SIP/2.0
  355. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK2bb1b5de;rport
  356. Max-Forwards: 70
  357. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  358. To: <sip:5124614444@trunk1.freepbx.com>
  359. Contact: <sip:6@71.244.49.87:5061>
  360. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  361. CSeq: 102 INVITE
  362. User-Agent: FPBX-13.0.120(13.7.1)
  363. Date: Wed, 01 Jun 2016 22:20:08 GMT
  364. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  365. Supported: replaces, timer
  366. Remote-Party-ID: "Cisco" <sip:6@71.244.49.87>;party=calling;privacy=off;screen=no
  367. Content-Type: application/sdp
  368. Content-Length: 251
  369.  
  370. v=0
  371. o=root 1617463576 1617463576 IN IP4 71.244.49.87
  372. s=Asterisk PBX 13.7.1
  373. c=IN IP4 71.244.49.87
  374. t=0 0
  375. m=audio 13292 RTP/AVP 0 101
  376. a=rtpmap:0 PCMU/8000
  377. a=rtpmap:101 telephone-event/8000
  378. a=fmtp:101 0-16
  379. a=ptime:20
  380. a=maxptime:150
  381. a=sendrecv
  382.  
  383. ---
  384. -- Called SIP/fpbx-1-cdB7e8PklPds/5124614444
  385.  
  386. <--- SIP read from UDP:192.159.66.3:5060 --->
  387. SIP/2.0 100 Trying
  388. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  389. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  390. To: <sip:5124614444@trunk1.freepbx.com>
  391. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  392. CSeq: 102 INVITE
  393. User-Agent: SIPStation 2.11.3
  394. Content-Length: 0
  395.  
  396. <------------->
  397. --- (8 headers 0 lines) ---
  398.  
  399. <--- SIP read from UDP:192.159.66.3:5060 --->
  400. SIP/2.0 407 Proxy Authentication Required
  401. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  402. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  403. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  404. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  405. CSeq: 102 INVITE
  406. User-Agent: SIPStation 2.11.3
  407. Accept: application/sdp
  408. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  409. Supported: timer, path, replaces
  410. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  411. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  412. Content-Length: 0
  413.  
  414. <------------->
  415. --- (13 headers 0 lines) ---
  416. Transmitting (NAT) to 192.159.66.3:5060:
  417. ACK sip:5124614444@trunk1.freepbx.com SIP/2.0
  418. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK2bb1b5de;rport
  419. Max-Forwards: 70
  420. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  421. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  422. Contact: <sip:6@71.244.49.87:5061>
  423. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  424. CSeq: 102 ACK
  425. User-Agent: FPBX-13.0.120(13.7.1)
  426. Content-Length: 0
  427.  
  428.  
  429. ---
  430. Audio is at 13292
  431. Adding codec ulaw to SDP
  432. Adding non-codec 0x1 (telephone-event) to SDP
  433. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  434. INVITE sip:5124614444@trunk1.freepbx.com SIP/2.0
  435. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK47323afc;rport
  436. Max-Forwards: 70
  437. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  438. To: <sip:5124614444@trunk1.freepbx.com>
  439. Contact: <sip:6@71.244.49.87:5061>
  440. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  441. CSeq: 103 INVITE
  442. User-Agent: FPBX-13.0.120(13.7.1)
  443. Proxy-Authorization: Digest username="cdB7e8PklPds", realm="71.244.49.87", algorithm=MD5, uri="sip:5124614444@trunk1.freepbx.com", nonce="fff58204-2846-11e6-af2e-0732f924a662", response="9b9e5872a04c0c8a87ebf13b2239b467", qop=auth, cnonce="586cb268", nc=00000001
  444. Date: Wed, 01 Jun 2016 22:20:08 GMT
  445. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  446. Supported: replaces, timer
  447. Remote-Party-ID: "Cisco" <sip:6@71.244.49.87>;party=calling;privacy=off;screen=no
  448. Content-Type: application/sdp
  449. Content-Length: 251
  450.  
  451. v=0
  452. o=root 1617463576 1617463577 IN IP4 71.244.49.87
  453. s=Asterisk PBX 13.7.1
  454. c=IN IP4 71.244.49.87
  455. t=0 0
  456. m=audio 13292 RTP/AVP 0 101
  457. a=rtpmap:0 PCMU/8000
  458. a=rtpmap:101 telephone-event/8000
  459. a=fmtp:101 0-16
  460. a=ptime:20
  461. a=maxptime:150
  462. a=sendrecv
  463.  
  464. ---
  465.  
  466. <--- SIP read from UDP:192.159.66.3:5060 --->
  467. SIP/2.0 100 Trying
  468. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK47323afc;rport=5061
  469. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  470. To: <sip:5124614444@trunk1.freepbx.com>
  471. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  472. CSeq: 103 INVITE
  473. User-Agent: SIPStation 2.11.3
  474. Content-Length: 0
  475.  
  476. <------------->
  477. --- (8 headers 0 lines) ---
  478. Reliably Transmitting (NAT) to 162.253.134.142:5060:
  479. OPTIONS sip:trunk2.freepbx.com SIP/2.0
  480. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK497b2a30;rport
  481. Max-Forwards: 70
  482. From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as0db99384
  483. To: <sip:trunk2.freepbx.com>
  484. Contact: <sip:Unknown@71.244.49.87:5061>
  485. Call-ID: 0ec87f166bda706753d0662f1ec51bfe@71.244.49.87:5061
  486. CSeq: 102 OPTIONS
  487. User-Agent: FPBX-13.0.120(13.7.1)
  488. Date: Wed, 01 Jun 2016 22:20:09 GMT
  489. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  490. Supported: replaces, timer
  491. Content-Length: 0
  492.  
  493.  
  494. ---
  495.  
  496. <--- SIP read from UDP:162.253.134.142:5060 --->
  497. SIP/2.0 200 OK
  498. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK497b2a30;rport=5061
  499. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as0db99384
  500. To: <sip:trunk2.freepbx.com>;tag=5cQ9NXe0Kg8FH
  501. Call-ID: 0ec87f166bda706753d0662f1ec51bfe@71.244.49.87:5061
  502. CSeq: 102 OPTIONS
  503. Contact: <sip:162.253.134.142>
  504. User-Agent: SIPStation 2.11.3
  505. Accept: application/sdp
  506. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  507. Supported: timer, path, replaces
  508. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  509. Content-Length: 0
  510.  
  511. <------------->
  512. --- (13 headers 0 lines) ---
  513. Really destroying SIP dialog '0ec87f166bda706753d0662f1ec51bfe@71.244.49.87:5061' Method: OPTIONS
  514. Reliably Transmitting (no NAT) to 192.168.1.170:5061:
  515. OPTIONS sip:6@192.168.1.170:5061 SIP/2.0
  516. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK6dd62926
  517. Max-Forwards: 70
  518. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as3899ba5a
  519. To: <sip:6@192.168.1.170:5061>
  520. Contact: <sip:Unknown@192.168.1.210:5061>
  521. Call-ID: 22cf9369165a1afa5668e7fc58d353ca@192.168.1.210:5061
  522. CSeq: 102 OPTIONS
  523. User-Agent: FPBX-13.0.120(13.7.1)
  524. Date: Wed, 01 Jun 2016 22:20:09 GMT
  525. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  526. Supported: replaces, timer
  527. Content-Length: 0
  528.  
  529.  
  530. ---
  531.  
  532. <--- SIP read from UDP:192.168.1.170:5061 --->
  533. SIP/2.0 200 OK
  534. To: <sip:6@192.168.1.170:5061>;tag=27d032d190b9c0d4i0
  535. From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as3899ba5a
  536. Call-ID: 22cf9369165a1afa5668e7fc58d353ca@192.168.1.210:5061
  537. CSeq: 102 OPTIONS
  538. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK6dd62926
  539. Server: Cisco/SPA501G-7.6.1
  540. Content-Length: 0
  541. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
  542. Supported: replaces
  543.  
  544. <------------->
  545. --- (10 headers 0 lines) ---
  546. Really destroying SIP dialog '22cf9369165a1afa5668e7fc58d353ca@192.168.1.210:5061' Method: OPTIONS
  547.  
  548. <--- SIP read from UDP:192.159.66.3:5060 --->
  549. SIP/2.0 407 Proxy Authentication Required
  550. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  551. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  552. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  553. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  554. CSeq: 102 INVITE
  555. User-Agent: SIPStation 2.11.3
  556. Accept: application/sdp
  557. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  558. Supported: timer, path, replaces
  559. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  560. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  561. Content-Length: 0
  562.  
  563. <------------->
  564. --- (13 headers 0 lines) ---
  565. Transmitting (NAT) to 192.159.66.3:5060:
  566. ACK sip:5124614444@trunk1.freepbx.com SIP/2.0
  567. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK47323afc;rport
  568. Max-Forwards: 70
  569. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  570. To: <sip:5124614444@trunk1.freepbx.com>
  571. Contact: <sip:6@71.244.49.87:5061>
  572. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  573. CSeq: 102 ACK
  574. User-Agent: FPBX-13.0.120(13.7.1)
  575. Content-Length: 0
  576.  
  577.  
  578. ---
  579. [2016-06-01 17:20:09] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:09.824-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49945",UsingPassword="0",SessionTV="2016-06-01T17:20:09.824-0500"
  580.  
  581. <--- SIP read from UDP:192.159.66.3:5060 --->
  582. SIP/2.0 407 Proxy Authentication Required
  583. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  584. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  585. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  586. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  587. CSeq: 102 INVITE
  588. User-Agent: SIPStation 2.11.3
  589. Accept: application/sdp
  590. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  591. Supported: timer, path, replaces
  592. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  593. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  594. Content-Length: 0
  595.  
  596. <------------->
  597. --- (13 headers 0 lines) ---
  598. Transmitting (NAT) to 192.159.66.3:5060:
  599. ACK sip:5124614444@trunk1.freepbx.com SIP/2.0
  600. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK47323afc;rport
  601. Max-Forwards: 70
  602. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  603. To: <sip:5124614444@trunk1.freepbx.com>
  604. Contact: <sip:6@71.244.49.87:5061>
  605. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  606. CSeq: 102 ACK
  607. User-Agent: FPBX-13.0.120(13.7.1)
  608. Content-Length: 0
  609.  
  610.  
  611. ---
  612.  
  613. <--- SIP read from UDP:192.159.66.3:5060 --->
  614. SIP/2.0 183 Session Progress
  615. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK47323afc;rport=5061
  616. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  617. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  618. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  619. CSeq: 103 INVITE
  620. Contact: <sip:5124614444@192.159.66.3:5060;transport=udp>
  621. User-Agent: SIPStation 2.11.3
  622. Accept: application/sdp
  623. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  624. Supported: timer, path, replaces
  625. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  626. Content-Type: application/sdp
  627. Content-Disposition: session
  628. Content-Length: 223
  629. Remote-Party-ID: "5124614444" <sip:5124614444@trunk1.freepbx.com>;party=calling;privacy=off;screen=no
  630.  
  631. v=0
  632. o=Sonus_UAC 769372 344197 IN IP4 67.231.13.80
  633. s=SIP Media Capabilities
  634. c=IN IP4 67.231.13.80
  635. t=0 0
  636. m=audio 63808 RTP/AVP 0 101
  637. a=rtpmap:0 PCMU/8000
  638. a=rtpmap:101 telephone-event/8000
  639. a=fmtp:101 0-15
  640. a=ptime:20
  641. <------------->
  642. --- (16 headers 10 lines) ---
  643. sip_route_dump: route/path hop: <sip:5124614444@192.159.66.3:5060;transport=udp>
  644. Found RTP audio format 0
  645. Found RTP audio format 101
  646. Found audio description format PCMU for ID 0
  647. Found audio description format telephone-event for ID 101
  648. Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  649. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  650. Peer audio RTP is at port 67.231.13.80:63808
  651. -- SIP/fpbx-1-cdB7e8PklPds-0000001d is making progress passing it to SIP/6-0000001c
  652. Audio is at 14978
  653. Adding codec ulaw to SDP
  654. Adding codec g722 to SDP
  655. Adding codec g729 to SDP
  656. Adding codec alaw to SDP
  657. Adding non-codec 0x1 (telephone-event) to SDP
  658.  
  659. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  660. SIP/2.0 183 Session Progress
  661. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-c41153e8;received=192.168.1.170
  662. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  663. To: <sip:5124614444@192.168.1.210>;tag=as5e82110d
  664. Call-ID: 8128a648-945b0518@192.168.1.170
  665. CSeq: 102 INVITE
  666. Server: FPBX-13.0.120(13.7.1)
  667. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  668. Supported: replaces, timer
  669. Contact: <sip:5124614444@192.168.1.210:5061>
  670. Content-Type: application/sdp
  671. Content-Length: 348
  672.  
  673. v=0
  674. o=root 1306462163 1306462163 IN IP4 192.168.1.210
  675. s=Asterisk PBX 13.7.1
  676. c=IN IP4 192.168.1.210
  677. t=0 0
  678. m=audio 14978 RTP/AVP 0 9 18 8 101
  679. a=rtpmap:0 PCMU/8000
  680. a=rtpmap:9 G722/8000
  681. a=rtpmap:18 G729/8000
  682. a=fmtp:18 annexb=no
  683. a=rtpmap:8 PCMA/8000
  684. a=rtpmap:101 telephone-event/8000
  685. a=fmtp:101 0-16
  686. a=ptime:20
  687. a=maxptime:150
  688. a=sendrecv
  689.  
  690. <------------>
  691. > 0x30a73e0 -- Probation passed - setting RTP source address to 192.168.1.170:16440
  692. [2016-06-01 17:20:12] NOTICE[1699]: chan_sip.c:15305 sip_reregister: -- Re-registration for cdB7e8PklPds@trunk2.freepbx.com
  693. REGISTER 12 headers, 0 lines
  694. Reliably Transmitting (NAT) to 162.253.134.142:5060:
  695. REGISTER sip:trunk2.freepbx.com SIP/2.0
  696. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6e09b0dd;rport
  697. Max-Forwards: 70
  698. From: <sip:cdB7e8PklPds@trunk2.freepbx.com>;tag=as15550875
  699. To: <sip:cdB7e8PklPds@trunk2.freepbx.com>
  700. Call-ID: 1bdf9d0e1c2f60f90595d3d33fd3d6b3@[::1]
  701. CSeq: 1082 REGISTER
  702. Supported: replaces, timer
  703. User-Agent: FPBX-13.0.120(13.7.1)
  704. Authorization: Digest username="cdB7e8PklPds", realm="trunk2.freepbx.com", algorithm=MD5, uri="sip:trunk2.freepbx.com", nonce="767a0cb0-2793-11e6-a689-3dbc13e3cded", response="be7b52f671de1c205dce0619355b8baf", qop=auth, cnonce="01531d7a", nc=000002df
  705. Expires: 120
  706. Contact: <sip:s@71.244.49.87:5061>
  707. Content-Length: 0
  708.  
  709.  
  710. ---
  711.  
  712. <--- SIP read from UDP:162.253.134.142:5060 --->
  713. SIP/2.0 200 OK
  714. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK6e09b0dd;rport=5061
  715. From: <sip:cdB7e8PklPds@trunk2.freepbx.com>;tag=as15550875
  716. To: <sip:cdB7e8PklPds@trunk2.freepbx.com>;tag=teeQyvvD5ZZHF
  717. Call-ID: 1bdf9d0e1c2f60f90595d3d33fd3d6b3@[::1]
  718. CSeq: 1082 REGISTER
  719. Contact: <sip:s@192.168.1.210:5061>;expires=120
  720. Date: Wed, 01 Jun 2016 22:20:12 GMT
  721. User-Agent: SIPStation 2.11.3
  722. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  723. Supported: timer, path, replaces
  724. Content-Length: 0
  725.  
  726. <------------->
  727. --- (12 headers 0 lines) ---
  728. [2016-06-01 17:20:12] NOTICE[1699]: chan_sip.c:23894 handle_response_register: Outbound Registration: Expiry for trunk2.freepbx.com is 120 sec (Scheduling reregistration in 105 s)
  729. Really destroying SIP dialog '1bdf9d0e1c2f60f90595d3d33fd3d6b3@[::1]' Method: REGISTER
  730.  
  731. <--- SIP read from UDP:192.159.66.3:5060 --->
  732. SIP/2.0 407 Proxy Authentication Required
  733. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  734. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  735. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  736. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  737. CSeq: 102 INVITE
  738. User-Agent: SIPStation 2.11.3
  739. Accept: application/sdp
  740. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  741. Supported: timer, path, replaces
  742. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  743. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  744. Content-Length: 0
  745.  
  746. <------------->
  747. --- (13 headers 0 lines) ---
  748. Transmitting (NAT) to 192.159.66.3:5060:
  749. ACK sip:5124614444@192.159.66.3:5060;transport=udp SIP/2.0
  750. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK47323afc;rport
  751. Max-Forwards: 70
  752. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  753. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  754. Contact: <sip:6@71.244.49.87:5061>
  755. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  756. CSeq: 102 ACK
  757. User-Agent: FPBX-13.0.120(13.7.1)
  758. Content-Length: 0
  759.  
  760.  
  761. ---
  762.  
  763. <--- SIP read from UDP:192.168.1.170:5061 --->
  764. NOTIFY sip:192.168.1.210:5061 SIP/2.0
  765. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-2f8135b9
  766. From: "Cisco" <sip:6@192.168.1.210>;tag=4e3052ad8cb6c2a8o0
  767. To: <sip:192.168.1.210>
  768. Call-ID: ec088d91-edff2514@192.168.1.170
  769. CSeq: 35977 NOTIFY
  770. Max-Forwards: 70
  771. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  772. Event: keep-alive
  773. User-Agent: Cisco/SPA501G-7.6.1
  774. Content-Length: 0
  775.  
  776. <------------->
  777. --- (11 headers 0 lines) ---
  778.  
  779. <--- Transmitting (NAT) to 192.168.1.170:5061 --->
  780. SIP/2.0 200 OK
  781. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-2f8135b9;received=192.168.1.170;rport=5061
  782. From: "Cisco" <sip:6@192.168.1.210>;tag=4e3052ad8cb6c2a8o0
  783. To: <sip:192.168.1.210>;tag=as3d1464ff
  784. Call-ID: ec088d91-edff2514@192.168.1.170
  785. CSeq: 35977 NOTIFY
  786. Server: FPBX-13.0.120(13.7.1)
  787. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  788. Supported: replaces, timer
  789. Content-Length: 0
  790.  
  791.  
  792. <------------>
  793. Scheduling destruction of SIP dialog 'ec088d91-edff2514@192.168.1.170' in 32000 ms (Method: NOTIFY)
  794. [2016-06-01 17:20:14] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:14.833-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49947",UsingPassword="0",SessionTV="2016-06-01T17:20:14.833-0500"
  795.  
  796. <--- SIP read from UDP:192.159.66.3:5060 --->
  797. SIP/2.0 200 OK
  798. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK47323afc;rport=5061
  799. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  800. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  801. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  802. CSeq: 103 INVITE
  803. Contact: <sip:5124614444@192.159.66.3:5060;transport=udp>
  804. User-Agent: SIPStation 2.11.3
  805. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  806. Supported: timer, path, replaces
  807. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  808. Content-Type: application/sdp
  809. Content-Disposition: session
  810. Content-Length: 223
  811. Remote-Party-ID: "Outbound Call" <sip:+15124614444@trunk1.freepbx.com>;party=calling;privacy=off;screen=no
  812.  
  813. v=0
  814. o=Sonus_UAC 769372 344197 IN IP4 67.231.13.80
  815. s=SIP Media Capabilities
  816. c=IN IP4 67.231.13.80
  817. t=0 0
  818. m=audio 63808 RTP/AVP 0 101
  819. a=rtpmap:0 PCMU/8000
  820. a=rtpmap:101 telephone-event/8000
  821. a=fmtp:101 0-15
  822. a=ptime:20
  823. <------------->
  824. --- (15 headers 10 lines) ---
  825. sip_route_dump: route/path hop: <sip:5124614444@192.159.66.3:5060;transport=udp>
  826. Transmitting (NAT) to 192.159.66.3:5060:
  827. ACK sip:5124614444@192.159.66.3:5060;transport=udp SIP/2.0
  828. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK7f181d63;rport
  829. Max-Forwards: 70
  830. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  831. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  832. Contact: <sip:6@71.244.49.87:5061>
  833. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  834. CSeq: 103 ACK
  835. User-Agent: FPBX-13.0.120(13.7.1)
  836. Content-Length: 0
  837.  
  838.  
  839. ---
  840. -- SIP/fpbx-1-cdB7e8PklPds-0000001d answered SIP/6-0000001c
  841. Audio is at 14978
  842. Adding codec ulaw to SDP
  843. Adding codec g722 to SDP
  844. Adding codec g729 to SDP
  845. Adding codec alaw to SDP
  846. Adding non-codec 0x1 (telephone-event) to SDP
  847.  
  848. <--- Reliably Transmitting (no NAT) to 192.168.1.170:5061 --->
  849. SIP/2.0 200 OK
  850. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-c41153e8;received=192.168.1.170
  851. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  852. To: <sip:5124614444@192.168.1.210>;tag=as5e82110d
  853. Call-ID: 8128a648-945b0518@192.168.1.170
  854. CSeq: 102 INVITE
  855. Server: FPBX-13.0.120(13.7.1)
  856. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  857. Supported: replaces, timer
  858. Contact: <sip:5124614444@192.168.1.210:5061>
  859. Content-Type: application/sdp
  860. Content-Length: 348
  861.  
  862. v=0
  863. o=root 1306462163 1306462163 IN IP4 192.168.1.210
  864. s=Asterisk PBX 13.7.1
  865. c=IN IP4 192.168.1.210
  866. t=0 0
  867. m=audio 14978 RTP/AVP 0 9 18 8 101
  868. a=rtpmap:0 PCMU/8000
  869. a=rtpmap:9 G722/8000
  870. a=rtpmap:18 G729/8000
  871. a=fmtp:18 annexb=no
  872. a=rtpmap:8 PCMA/8000
  873. a=rtpmap:101 telephone-event/8000
  874. a=fmtp:101 0-16
  875. a=ptime:20
  876. a=maxptime:150
  877. a=sendrecv
  878.  
  879. <------------>
  880. -- Channel SIP/fpbx-1-cdB7e8PklPds-0000001d joined 'simple_bridge' basic-bridge <314c1605-25ab-43a5-ba6a-39b2f72a20c4>
  881. -- Channel SIP/6-0000001c joined 'simple_bridge' basic-bridge <314c1605-25ab-43a5-ba6a-39b2f72a20c4>
  882.  
  883. <--- SIP read from UDP:192.168.1.170:5061 --->
  884. ACK sip:5124614444@192.168.1.210:5061 SIP/2.0
  885. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-f281d702
  886. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  887. To: <sip:5124614444@192.168.1.210>;tag=as5e82110d
  888. Call-ID: 8128a648-945b0518@192.168.1.170
  889. CSeq: 102 ACK
  890. Max-Forwards: 70
  891. Authorization: Digest username="6",realm="asterisk",nonce="23998573",uri="sip:5124614444@192.168.1.210:5061",algorithm=MD5,response="13b76bdbf26951c3a4a55a42458975ab"
  892. Contact: "Cisco" <sip:6@192.168.1.170:5061>
  893. User-Agent: Cisco/SPA501G-7.6.1
  894. Content-Length: 0
  895.  
  896. <------------->
  897. --- (11 headers 0 lines) ---
  898.  
  899. <--- SIP read from UDP:192.159.66.3:5060 --->
  900. SIP/2.0 407 Proxy Authentication Required
  901. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  902. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  903. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  904. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  905. CSeq: 102 INVITE
  906. User-Agent: SIPStation 2.11.3
  907. Accept: application/sdp
  908. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  909. Supported: timer, path, replaces
  910. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  911. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  912. Content-Length: 0
  913.  
  914. <------------->
  915. --- (13 headers 0 lines) ---
  916. [2016-06-01 17:20:19] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:19.825-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49949",UsingPassword="0",SessionTV="2016-06-01T17:20:19.825-0500"
  917.  
  918. <--- SIP read from UDP:192.159.66.3:5060 --->
  919. SIP/2.0 407 Proxy Authentication Required
  920. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  921. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  922. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  923. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  924. CSeq: 102 INVITE
  925. User-Agent: SIPStation 2.11.3
  926. Accept: application/sdp
  927. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  928. Supported: timer, path, replaces
  929. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  930. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  931. Content-Length: 0
  932.  
  933. <------------->
  934. --- (13 headers 0 lines) ---
  935.  
  936. <--- SIP read from UDP:192.168.1.170:5061 --->
  937. BYE sip:5124614444@192.168.1.210:5061 SIP/2.0
  938. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-82000198
  939. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  940. To: <sip:5124614444@192.168.1.210>;tag=as5e82110d
  941. Call-ID: 8128a648-945b0518@192.168.1.170
  942. CSeq: 103 BYE
  943. Max-Forwards: 70
  944. Authorization: Digest username="6",realm="asterisk",nonce="23998573",uri="sip:5124614444@192.168.1.210:5061",algorithm=MD5,response="5a88b871a12ff79b5af280f501cffc90"
  945. User-Agent: Cisco/SPA501G-7.6.1
  946. Content-Length: 0
  947.  
  948. <------------->
  949. --- (10 headers 0 lines) ---
  950. Sending to 192.168.1.170:5061 (no NAT)
  951. Scheduling destruction of SIP dialog '8128a648-945b0518@192.168.1.170' in 6400 ms (Method: BYE)
  952.  
  953. <--- Transmitting (no NAT) to 192.168.1.170:5061 --->
  954. SIP/2.0 200 OK
  955. Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-82000198;received=192.168.1.170
  956. From: "Cisco" <sip:6@192.168.1.210>;tag=4807b488303d4bd8o0
  957. To: <sip:5124614444@192.168.1.210>;tag=as5e82110d
  958. Call-ID: 8128a648-945b0518@192.168.1.170
  959. CSeq: 103 BYE
  960. Server: FPBX-13.0.120(13.7.1)
  961. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  962. Supported: replaces, timer
  963. Content-Length: 0
  964.  
  965.  
  966. <------------>
  967. -- Channel SIP/6-0000001c left 'simple_bridge' basic-bridge <314c1605-25ab-43a5-ba6a-39b2f72a20c4>
  968. -- Channel SIP/fpbx-1-cdB7e8PklPds-0000001d left 'simple_bridge' basic-bridge <314c1605-25ab-43a5-ba6a-39b2f72a20c4>
  969. Scheduling destruction of SIP dialog '2efd64e3361011e95705ae07326271e0@71.244.49.87:5061' in 6400 ms (Method: INVITE)
  970. Reliably Transmitting (NAT) to 192.159.66.3:5060:
  971. BYE sip:5124614444@192.159.66.3:5060;transport=udp SIP/2.0
  972. Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK3fd44194;rport
  973. Max-Forwards: 70
  974. From: "Cisco" <sip:6@71.244.49.87:5061>;tag=as090cf3c7
  975. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  976. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  977. CSeq: 104 BYE
  978. User-Agent: FPBX-13.0.120(13.7.1)
  979. Proxy-Authorization: Digest username="cdB7e8PklPds", realm="71.244.49.87", algorithm=MD5, uri="sip:5124614444@192.159.66.3:5060", nonce="fff58204-2846-11e6-af2e-0732f924a662", response="9826e876fefc60cc972892fbe4d93471", qop=auth, cnonce="323a07b5", nc=00000002
  980. X-Asterisk-HangupCause: Normal Clearing
  981. X-Asterisk-HangupCauseCode: 16
  982. Content-Length: 0
  983.  
  984.  
  985. ---
  986. == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/6-0000001c' in macro 'dialout-trunk'
  987. == Spawn extension (restrictedroute-98bd5f7b1447e8791389136169a3a580, 5124614444, 6) exited non-zero on 'SIP/6-0000001c'
  988. -- Executing [h@restrictedroute-98bd5f7b1447e8791389136169a3a580:1] Hangup("SIP/6-0000001c", "") in new stack
  989. == Spawn extension (restrictedroute-98bd5f7b1447e8791389136169a3a580, h, 1) exited non-zero on 'SIP/6-0000001c'
  990.  
  991. <--- SIP read from UDP:192.159.66.3:5060 --->
  992. SIP/2.0 200 OK
  993. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK3fd44194;rport=5061
  994. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  995. To: <sip:5124614444@trunk1.freepbx.com>;tag=K1974QS0vZ3jN
  996. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  997. CSeq: 104 BYE
  998. User-Agent: SIPStation 2.11.3
  999. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  1000. Supported: timer, path, replaces
  1001. Content-Length: 0
  1002.  
  1003. <------------->
  1004. --- (10 headers 0 lines) ---
  1005. Really destroying SIP dialog '2efd64e3361011e95705ae07326271e0@71.244.49.87:5061' Method: INVITE
  1006.  
  1007. <--- SIP read from UDP:192.159.66.3:5060 --->
  1008. SIP/2.0 407 Proxy Authentication Required
  1009. Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK2bb1b5de;rport=5061
  1010. From: "Cisco" <sip:6@192.168.1.210:5061>;tag=as090cf3c7
  1011. To: <sip:5124614444@trunk1.freepbx.com>;tag=jrgF3v8vZpD0S
  1012. Call-ID: 2efd64e3361011e95705ae07326271e0@71.244.49.87:5061
  1013. CSeq: 102 INVITE
  1014. User-Agent: SIPStation 2.11.3
  1015. Accept: application/sdp
  1016. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  1017. Supported: timer, path, replaces
  1018. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  1019. Proxy-Authenticate: Digest realm="71.244.49.87", nonce="fff58204-2846-11e6-af2e-0732f924a662", algorithm=MD5, qop="auth"
  1020. Content-Length: 0
  1021.  
  1022. <------------->
  1023. --- (13 headers 0 lines) ---
  1024. [2016-06-01 17:20:24] SECURITY[1769]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-06-01T17:20:24.831-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x2e76338",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/49951",UsingPassword="0",SessionTV="2016-06-01T17:20:24.831-0500"
  1025. localhost*CLI>
  1026. Disconnected from Asterisk server
  1027. Asterisk cleanly ending (0).
  1028. Executing last minute cleanups
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