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  1. Audio is at 96.9.137.12 port 12188
  2. Adding codec 0x4 (ulaw) to SDP
  3. Adding non-codec 0x1 (telephone-event) to SDP
  4. Reliably Transmitting (NAT) to 70.167.153.130:5060:
  5. INVITE sip:19096614521@sip.flowroute.com SIP/2.0
  6. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK4132b74f;rport
  7. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  8. To: <sip:19096614521@sip.flowroute.com>
  9. Contact: <sip:225@96.9.137.12>
  10. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  11. CSeq: 102 INVITE
  12. User-Agent: Asterisk PBX
  13. Max-Forwards: 70
  14. Remote-Party-ID: "225" <sip:225@sip.flowroute.com>;privacy=off;screen=no
  15. Date: Tue, 16 Feb 2010 20:34:26 GMT
  16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  17. Supported: replaces
  18. Content-Type: application/sdp
  19. Content-Length: 238
  20.  
  21. v=0
  22. o=root 28194 28194 IN IP4 96.9.137.12
  23. s=session
  24. c=IN IP4 96.9.137.12
  25. t=0 0
  26. m=audio 12188 RTP/AVP 0 101
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:101 telephone-event/8000
  29. a=fmtp:101 0-16
  30. a=silenceSupp:off - - - -
  31. a=ptime:20
  32. a=sendrecv
  33.  
  34. ---
  35. -- Called flowroute/19096614521
  36. webcd*CLI>
  37. <--- SIP read from 70.167.153.130:5060 --->
  38. SIP/2.0 100 Trying
  39. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK4132b74f;rport=5060
  40. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  41. To: <sip:19096614521@sip.flowroute.com>
  42. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  43. CSeq: 102 INVITE
  44. Content-Length: 0
  45.  
  46.  
  47. <------------->
  48. --- (7 headers 0 lines) ---
  49. webcd*CLI>
  50. <--- SIP read from 70.167.153.130:5060 --->
  51. SIP/2.0 407 Proxy Authentication Required
  52. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK4132b74f;rport=5060
  53. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  54. To: <sip:19096614521@sip.flowroute.com>;tag=b119c633111171d5a41b4118046e5815.2d95
  55. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  56. CSeq: 102 INVITE
  57. Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="4b7b01030001776bdc5d638cdc55f3bd62cb0e9eb2ad820a", qop="auth"
  58. Content-Length: 0
  59.  
  60.  
  61. <------------->
  62. --- (8 headers 0 lines) ---
  63. Transmitting (NAT) to 70.167.153.130:5060:
  64. ACK sip:19096614521@sip.flowroute.com SIP/2.0
  65. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK4132b74f;rport
  66. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  67. To: <sip:19096614521@sip.flowroute.com>;tag=b119c633111171d5a41b4118046e5815.2d95
  68. Contact: <sip:225@96.9.137.12>
  69. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  70. CSeq: 102 ACK
  71. User-Agent: Asterisk PBX
  72. Max-Forwards: 70
  73. Remote-Party-ID: "225" <sip:225@sip.flowroute.com>;privacy=off;screen=no
  74. Content-Length: 0
  75.  
  76.  
  77. ---
  78. Audio is at 96.9.137.12 port 12188
  79. Adding codec 0x4 (ulaw) to SDP
  80. Adding non-codec 0x1 (telephone-event) to SDP
  81. Reliably Transmitting (NAT) to 70.167.153.130:5060:
  82. INVITE sip:19096614521@sip.flowroute.com SIP/2.0
  83. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK793e675d;rport
  84. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  85. To: <sip:19096614521@sip.flowroute.com>
  86. Contact: <sip:225@96.9.137.12>
  87. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  88. CSeq: 103 INVITE
  89. User-Agent: Asterisk PBX
  90. Max-Forwards: 70
  91. Remote-Party-ID: "225" <sip:225@sip.flowroute.com>;privacy=off;screen=no
  92. Proxy-Authorization: Digest username="35097357", realm="sip.flowroute.com", algorithm=MD5, uri="sip:19096614521@sip.flowroute.com", nonce="4b7b01030001776bdc5d638cdc55f3bd62cb0e9eb2ad820a", response="84cb2546980f4a94abadb7e5f8a21f2c", qop=auth, cnonce="7b737618", nc=00000001
  93. Date: Tue, 16 Feb 2010 20:34:26 GMT
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  95. Supported: replaces
  96. Content-Type: application/sdp
  97. Content-Length: 238
  98.  
  99. v=0
  100. o=root 28194 28195 IN IP4 96.9.137.12
  101. s=session
  102. c=IN IP4 96.9.137.12
  103. t=0 0
  104. m=audio 12188 RTP/AVP 0 101
  105. a=rtpmap:0 PCMU/8000
  106. a=rtpmap:101 telephone-event/8000
  107. a=fmtp:101 0-16
  108. a=silenceSupp:off - - - -
  109. a=ptime:20
  110. a=sendrecv
  111.  
  112. ---
  113. webcd*CLI>
  114. <--- SIP read from 70.167.153.130:5060 --->
  115. SIP/2.0 100 Trying
  116. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK793e675d;rport=5060
  117. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  118. To: <sip:19096614521@sip.flowroute.com>
  119. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  120. CSeq: 103 INVITE
  121. Content-Length: 0
  122.  
  123.  
  124. <------------->
  125. --- (7 headers 0 lines) ---
  126. webcd*CLI>
  127. <--- SIP read from 70.167.153.130:5060 --->
  128. SIP/2.0 183 Session Progress
  129. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK793e675d;rport=5060
  130. From: <sip:225@flowroute.com>;tag=as27c7df4d
  131. To: <sip:9096614521@flowroute.com>;tag=14c02ea8+1+63b10094+f7e27ead
  132. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  133. CSeq: 103 INVITE
  134. Record-Route: <sip:216.115.69.133;lr>
  135. Record-Route: <sip:70.167.153.130;lr>
  136. Contact: <sip:9096614521@65.98.237.158:5060;npdi=yes;transport=udp>
  137. Content-Type: application/sdp
  138. Content-Length: 200
  139.  
  140. v=0
  141. o=- 3475341160 3475341176 IN IP4 65.98.237.158
  142. s=-
  143. c=IN IP4 65.98.237.158
  144. t=0 0
  145. m=audio 13250 RTP/AVP 0 101
  146. a=rtpmap:101 telephone-event/8000
  147. a=fmtp:101 0-15
  148. a=ptime:20
  149. a=direction:both
  150.  
  151. <------------->
  152. --- (11 headers 10 lines) ---
  153. Found RTP audio format 0
  154. Found RTP audio format 101
  155. Found audio description format telephone-event for ID 101
  156. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  157. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  158. Peer audio RTP is at port 65.98.237.158:13250
  159. -- SIP/flowroute-00000041 is making progress passing it to SIP/225-00000040
  160. webcd*CLI>
  161. <--- SIP read from 70.167.153.130:5060 --->
  162. SIP/2.0 200 OK
  163. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK793e675d;rport=5060
  164. From: <sip:225@flowroute.com>;tag=as27c7df4d
  165. To: <sip:9096614521@flowroute.com>;tag=14c02ea8+1+63b10094+f7e27ead
  166. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  167. CSeq: 103 INVITE
  168. Record-Route: <sip:216.115.69.133;lr>
  169. Record-Route: <sip:70.167.153.130;lr>
  170. Contact: <sip:9096614521@65.98.237.158:5060;npdi=yes;transport=udp>
  171. Content-Type: application/sdp
  172. Content-Length: 200
  173.  
  174. v=0
  175. o=- 3475341160 3475341176 IN IP4 65.98.237.158
  176. s=-
  177. c=IN IP4 65.98.237.158
  178. t=0 0
  179. m=audio 13250 RTP/AVP 0 101
  180. a=rtpmap:101 telephone-event/8000
  181. a=fmtp:101 0-15
  182. a=ptime:20
  183. a=direction:both
  184.  
  185. <------------->
  186. --- (11 headers 10 lines) ---
  187. Found RTP audio format 0
  188. Found RTP audio format 101
  189. Found audio description format telephone-event for ID 101
  190. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  191. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  192. Peer audio RTP is at port 65.98.237.158:13250
  193. list_route: hop: <sip:70.167.153.130;lr>
  194. list_route: hop: <sip:216.115.69.133;lr>
  195. set_destination: Parsing <sip:70.167.153.130;lr> for address/port to send to
  196. set_destination: set destination to 70.167.153.130, port 5060
  197. Transmitting (NAT) to 70.167.153.130:5060:
  198. ACK sip:9096614521@65.98.237.158:5060;npdi=yes;transport=udp SIP/2.0
  199. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK2149a094;rport
  200. Route: <sip:70.167.153.130;lr>,<sip:216.115.69.133;lr>
  201. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  202. To: <sip:19096614521@sip.flowroute.com>;tag=14c02ea8+1+63b10094+f7e27ead
  203. Contact: <sip:225@96.9.137.12>
  204. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  205. CSeq: 103 ACK
  206. User-Agent: Asterisk PBX
  207. Max-Forwards: 70
  208. Remote-Party-ID: "225" <sip:225@sip.flowroute.com>;privacy=off;screen=no
  209. Content-Length: 0
  210.  
  211.  
  212. ---
  213. -- SIP/flowroute-00000041 answered SIP/225-00000040
  214. -- Packet2Packet bridging SIP/225-00000040 and SIP/flowroute-00000041
  215. webcd*CLI>
  216. webcd*CLI>
  217. webcd*CLI>
  218. webcd*CLI>
  219. webcd*CLI>
  220. webcd*CLI>
  221. webcd*CLI>
  222. webcd*CLI>
  223. webcd*CLI>
  224. webcd*CLI>
  225. webcd*CLI>
  226. REGISTER 13 headers, 0 lines
  227. Reliably Transmitting (no NAT) to 70.167.153.130:5060:
  228. REGISTER sip:sip.flowroute.com SIP/2.0
  229. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK2ed44e79;rport
  230. From: <sip:35097357@sip.flowroute.com>;tag=as1b76392d
  231. To: <sip:35097357@sip.flowroute.com>
  232. Call-ID: 4b94108579e3e21b098882611f39c63a@96.9.137.12
  233. CSeq: 202 REGISTER
  234. User-Agent: Asterisk PBX
  235. Max-Forwards: 70
  236. Authorization: Digest username="35097357", realm="sip.flowroute.com", algorithm=MD5, uri="sip:sip.flowroute.com", nonce="4b7b009f000138db27e5b4af57f3f4bc8d3c68bfaa072a71", response="65c4405b957051dead0f94383b1fc0bd", qop=auth, cnonce="61efbe3d", nc=00000002
  237. Expires: 120
  238. Contact: <sip:s@96.9.137.12>
  239. Event: registration
  240. Content-Length: 0
  241.  
  242.  
  243. ---
  244. webcd*CLI>
  245. <--- SIP read from 70.167.153.130:5060 --->
  246. SIP/2.0 401 Unauthorized
  247. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK2ed44e79;rport=5060
  248. From: <sip:35097357@sip.flowroute.com>;tag=as1b76392d
  249. To: <sip:35097357@sip.flowroute.com>;tag=3ac9d3045d396da2a5a9a811a285b36c.4754
  250. Call-ID: 4b94108579e3e21b098882611f39c63a@96.9.137.12
  251. CSeq: 202 REGISTER
  252. WWW-Authenticate: Digest realm="sip.flowroute.com", nonce="4b7b010900013d785da736431413eb876bd45a3281eb9624", qop="auth", stale=true
  253. Content-Length: 0
  254.  
  255.  
  256. <------------->
  257. --- (8 headers 0 lines) ---
  258. Responding to challenge, registration to domain/host name sip.flowroute.com
  259. REGISTER 13 headers, 0 lines
  260. Reliably Transmitting (no NAT) to 70.167.153.130:5060:
  261. REGISTER sip:sip.flowroute.com SIP/2.0
  262. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK710832d4;rport
  263. From: <sip:35097357@sip.flowroute.com>;tag=as18f325dc
  264. To: <sip:35097357@sip.flowroute.com>
  265. Call-ID: 4b94108579e3e21b098882611f39c63a@96.9.137.12
  266. CSeq: 203 REGISTER
  267. User-Agent: Asterisk PBX
  268. Max-Forwards: 70
  269. Authorization: Digest username="35097357", realm="sip.flowroute.com", algorithm=MD5, uri="sip:sip.flowroute.com", nonce="4b7b010900013d785da736431413eb876bd45a3281eb9624", response="b1fac214ee51bdda4d7291c82b6fbc9b", qop=auth, cnonce="4a85ae7b", nc=00000001
  270. Expires: 120
  271. Contact: <sip:s@96.9.137.12>
  272. Event: registration
  273. Content-Length: 0
  274.  
  275.  
  276. ---
  277. webcd*CLI>
  278. <--- SIP read from 70.167.153.130:5060 --->
  279. SIP/2.0 200 OK
  280. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK710832d4;rport=5060
  281. From: <sip:35097357@sip.flowroute.com>;tag=as18f325dc
  282. To: <sip:35097357@sip.flowroute.com>;tag=3ac9d3045d396da2a5a9a811a285b36c.76a7
  283. Call-ID: 4b94108579e3e21b098882611f39c63a@96.9.137.12
  284. CSeq: 203 REGISTER
  285. Contact: <sip:s@96.9.137.12>;q=1;expires=120;received="sip:96.9.137.12:5060"
  286. Content-Length: 0
  287.  
  288.  
  289. <------------->
  290. --- (8 headers 0 lines) ---
  291. Scheduling destruction of SIP dialog '4b94108579e3e21b098882611f39c63a@96.9.137.12' in 32000 ms (Method: REGISTER)
  292. webcd*CLI>
  293. <--- SIP read from 70.167.153.130:5060 --->
  294. OPTIONS sip:96.9.137.12:5060 SIP/2.0
  295. Max-Forwards: 10
  296. Record-Route: <sip:70.167.153.130;lr>
  297. Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK2154.d8c3e9371d2d3666f668e78a2cdc5664.0
  298. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  299. Route: <sip:70.167.153.130;lr;received="sip:96.9.137.12:5060">
  300. From: sip:ping@invalid;tag=210a6927
  301. To: sip:96.9.137.12:5060
  302. Call-ID: a51e63d1-4f931124-b16852@216.115.69.131
  303. CSeq: 1 OPTIONS
  304. Content-Length: 0
  305.  
  306.  
  307. <------------->
  308. --- (11 headers 0 lines) ---
  309. Looking for s in lawl (domain 96.9.137.12)
  310. webcd*CLI>
  311. <--- Transmitting (no NAT) to 70.167.153.130:5060 --->
  312. SIP/2.0 404 Not Found
  313. Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK2154.d8c3e9371d2d3666f668e78a2cdc5664.0;received=70.167.153.130
  314. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  315. From: sip:ping@invalid;tag=210a6927
  316. To: sip:96.9.137.12:5060;tag=as457468f1
  317. Call-ID: a51e63d1-4f931124-b16852@216.115.69.131
  318. CSeq: 1 OPTIONS
  319. User-Agent: Asterisk PBX
  320. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  321. Supported: replaces
  322. Accept: application/sdp
  323. Content-Length: 0
  324.  
  325.  
  326.  
  327.  
  328. SO FAR THE CALL IS FINE.
  329.  
  330.  
  331. 30 seconds later
  332.  
  333.  
  334.  
  335. the call drops
  336.  
  337. Scheduling destruction of SIP dialog '4eb065131899ed777a895b8868beb7dd@sip.flowroute.com' in 32000 ms (Method: INVITE)
  338. set_destination: Parsing <sip:70.167.153.130;lr> for address/port to send to
  339. set_destination: set destination to 70.167.153.130, port 5060
  340. Reliably Transmitting (NAT) to 70.167.153.130:5060:
  341. BYE sip:9096614521@65.98.237.158:5060;npdi=yes;transport=udp SIP/2.0
  342. Via: SIP/2.0/UDP 96.9.137.12:5060;branch=z9hG4bK7007134d;rport
  343. Route: <sip:70.167.153.130;lr>,<sip:216.115.69.133;lr>
  344. From: "225" <sip:225@sip.flowroute.com>;tag=as27c7df4d
  345. To: <sip:19096614521@sip.flowroute.com>;tag=14c02ea8+1+63b10094+f7e27ead
  346. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  347. CSeq: 104 BYE
  348. User-Agent: Asterisk PBX
  349. Max-Forwards: 70
  350. Remote-Party-ID: "225" <sip:225@sip.flowroute.com>;privacy=off;screen=no
  351. Proxy-Authorization: Digest username="35097357", realm="sip.flowroute.com", algorithm=MD5, uri="sip:9096614521@65.98.237.158:5060", nonce="4b7b01030001776bdc5d638cdc55f3bd62cb0e9eb2ad820a", response="3b7d1ab4e456a87fe6870ef7c6a0d0ff", qop=auth, cnonce="117ed100", nc=00000002
  352. X-Asterisk-HangupCause: Normal Clearing
  353. X-Asterisk-HangupCauseCode: 16
  354. Content-Length: 0
  355.  
  356.  
  357. ---
  358. == Spawn extension (lucky, 19096614521, 5) exited non-zero on 'SIP/225-00000040'
  359. webcd*CLI>
  360. <--- SIP read from 70.167.153.130:5060 --->
  361. SIP/2.0 200 OK
  362. Via: SIP/2.0/UDP 96.9.137.12:5060;received=96.9.137.12;branch=z9hG4bK7007134d;rport=5060
  363. From: <sip:225@flowroute.com>;tag=as27c7df4d
  364. To: <sip:9096614521@flowroute.com>;tag=14c02ea8+1+63b10094+f7e27ead
  365. Call-ID: 4eb065131899ed777a895b8868beb7dd@sip.flowroute.com
  366. CSeq: 104 BYE
  367. Content-Length: 0
  368.  
  369.  
  370. <------------->
  371. --- (7 headers 0 lines) ---
  372. Really destroying SIP dialog '4eb065131899ed777a895b8868beb7dd@sip.flowroute.com' Method: INVITE
  373. Really destroying SIP dialog 'OTE3ZjEyN2FkYTU4NWYwZmNjOWY5M2ExMjk5ZWVhOGQ.' Method: BYE
  374. Really destroying SIP dialog '4b94108579e3e21b098882611f39c63a@96.9.137.12' Method: REGISTER
  375. Really destroying SIP dialog 'a51e63d1-4f931124-b16852@216.115.69.131' Method: OPTIONS
  376. Really destroying SIP dialog '6c7332b67c086c4b123ffc2d0ed44c0d@96.9.137.12' Method: OPTIONS
  377. Really destroying SIP dialog 'NmQ0NzIzNGZjOTJkNDBlMzVkODI0Y2VhMjQ5MTk3Njk.' Method: SUBSCRIBE
  378. webcd*CLI>
  379. <--- SIP read from 70.167.153.130:5060 --->
  380. OPTIONS sip:96.9.137.12:5060 SIP/2.0
  381. Max-Forwards: 10
  382. Record-Route: <sip:70.167.153.130;lr>
  383. Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKc306.0e6f4cbe3e6ef0cf9afcdab5e50e39d0.0
  384. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  385. Route: <sip:70.167.153.130;lr;received="sip:96.9.137.12:5060">
  386. From: sip:ping@invalid;tag=6b4a6927
  387. To: sip:96.9.137.12:5060
  388. Call-ID: a51e63d1-89e31124-256852@216.115.69.131
  389. CSeq: 1 OPTIONS
  390. Content-Length: 0
  391.  
  392.  
  393. <------------->
  394. --- (11 headers 0 lines) ---
  395. Looking for s in lawl (domain 96.9.137.12)
  396. webcd*CLI>
  397. <--- Transmitting (no NAT) to 70.167.153.130:5060 --->
  398. SIP/2.0 404 Not Found
  399. Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKc306.0e6f4cbe3e6ef0cf9afcdab5e50e39d0.0;received=70.167.153.130
  400. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  401. From: sip:ping@invalid;tag=6b4a6927
  402. To: sip:96.9.137.12:5060;tag=as78c137d1
  403. Call-ID: a51e63d1-89e31124-256852@216.115.69.131
  404. CSeq: 1 OPTIONS
  405. User-Agent: Asterisk PBX
  406. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  407. Supported: replaces
  408. Accept: application/sdp
  409. Content-Length: 0
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