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Aug 23rd, 2016
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  1.  
  2. ---
  3.  
  4. <--- SIP read from UDP:183.76.169.117:40408 --->
  5. SIP/2.0 200 OK
  6. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK6d40167d;rport=5060
  7. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5aaa9e92
  8. To: <sip:200@10.0.1.34:5060>;tag=159625396
  9. Call-ID: 757874571e032eb02655b4475cc1b875@110.5.42.156:5060
  10. CSeq: 102 OPTIONS
  11. Supported: replaces, path, timer
  12. User-Agent: Grandstream GXP1625 1.0.2.27
  13. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  14. Content-Length: 0
  15.  
  16. <------------->
  17. --- (10 headers 0 lines) ---
  18. Really destroying SIP dialog '757874571e032eb02655b4475cc1b875@110.5.42.156:5060' Method: OPTIONS
  19. localhost*CLI> core set verbose 10
  20. Console verbose was 1 and is now 10.
  21. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  22. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  23. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK37fd851e;rport
  24. Max-Forwards: 70
  25. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4cfd84f9
  26. To: <sip:200@10.0.1.34:5060>
  27. Contact: <sip:asterisk@110.5.42.156:5060>
  28. Call-ID: 5786198e22659092152b373506ab10bc@110.5.42.156:5060
  29. CSeq: 102 OPTIONS
  30. User-Agent: Asterisk PBX 11.23.0
  31. Date: Tue, 23 Aug 2016 11:04:03 GMT
  32. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  33. Supported: replaces, timer
  34. Content-Length: 0
  35.  
  36.  
  37. ---
  38.  
  39. <--- SIP read from UDP:183.76.169.117:40408 --->
  40. SIP/2.0 200 OK
  41. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK37fd851e;rport=5060
  42. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4cfd84f9
  43. To: <sip:200@10.0.1.34:5060>;tag=289265956
  44. Call-ID: 5786198e22659092152b373506ab10bc@110.5.42.156:5060
  45. CSeq: 102 OPTIONS
  46. Supported: replaces, path, timer
  47. User-Agent: Grandstream GXP1625 1.0.2.27
  48. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  49. Content-Length: 0
  50.  
  51. <------------->
  52. --- (10 headers 0 lines) ---
  53. Really destroying SIP dialog '5786198e22659092152b373506ab10bc@110.5.42.156:5060' Method: OPTIONS
  54. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  55. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  56. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK4ac6ba07;rport
  57. Max-Forwards: 70
  58. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as76d16ca5
  59. To: <sip:200@10.0.1.34:5060>
  60. Contact: <sip:asterisk@110.5.42.156:5060>
  61. Call-ID: 74f40f22646ca8ae37c990cd7d15fb22@110.5.42.156:5060
  62. CSeq: 102 OPTIONS
  63. User-Agent: Asterisk PBX 11.23.0
  64. Date: Tue, 23 Aug 2016 11:04:23 GMT
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  66. Supported: replaces, timer
  67. Content-Length: 0
  68.  
  69.  
  70. ---
  71.  
  72. <--- SIP read from UDP:183.76.169.117:40408 --->
  73. SIP/2.0 200 OK
  74. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK4ac6ba07;rport=5060
  75. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as76d16ca5
  76. To: <sip:200@10.0.1.34:5060>;tag=1146002030
  77. Call-ID: 74f40f22646ca8ae37c990cd7d15fb22@110.5.42.156:5060
  78. CSeq: 102 OPTIONS
  79. Supported: replaces, path, timer
  80. User-Agent: Grandstream GXP1625 1.0.2.27
  81. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  82. Content-Length: 0
  83.  
  84. <------------->
  85. --- (10 headers 0 lines) ---
  86. Really destroying SIP dialog '74f40f22646ca8ae37c990cd7d15fb22@110.5.42.156:5060' Method: OPTIONS
  87.  
  88. <--- SIP read from UDP:183.76.169.117:40408 --->
  89. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  90. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1348139891;rport
  91. From: <sip:200@110.5.42.156>;tag=158908947
  92. To: <sip:09016192354@110.5.42.156>
  93. Call-ID: 1307273081-5060-21@BA.A.B.DE
  94. CSeq: 200 INVITE
  95. Contact: <sip:200@10.0.1.34:5060>
  96. Max-Forwards: 70
  97. User-Agent: Grandstream GXP1625 1.0.2.27
  98. Privacy: none
  99. P-Preferred-Identity: <sip:200@110.5.42.156>
  100. Supported: replaces, path, timer
  101. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  102. Content-Type: application/sdp
  103. Accept: application/sdp, application/dtmf-relay
  104. Content-Length: 326
  105.  
  106. v=0
  107. o=200 8000 8000 IN IP4 10.0.1.34
  108. s=SIP Call
  109. c=IN IP4 10.0.1.34
  110. t=0 0
  111. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  112. a=sendrecv
  113. a=rtpmap:0 PCMU/8000
  114. a=ptime:20
  115. a=rtpmap:8 PCMA/8000
  116. a=rtpmap:18 G729/8000
  117. a=fmtp:18 annexb=no
  118. a=rtpmap:9 G722/8000
  119. a=rtpmap:2 G726-32/8000
  120. a=rtpmap:101 telephone-event/8000
  121. a=fmtp:101 0-15
  122. <------------->
  123. --- (16 headers 16 lines) ---
  124. Sending to 183.76.169.117:40408 (NAT)
  125. Sending to 183.76.169.117:40408 (NAT)
  126. Using INVITE request as basis request - 1307273081-5060-21@BA.A.B.DE
  127. Found peer '200' for '200' from 183.76.169.117:40408
  128.  
  129. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  130. SIP/2.0 401 Unauthorized
  131. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1348139891;received=183.76.169.117;rport=40408
  132. From: <sip:200@110.5.42.156>;tag=158908947
  133. To: <sip:09016192354@110.5.42.156>;tag=as10029f86
  134. Call-ID: 1307273081-5060-21@BA.A.B.DE
  135. CSeq: 200 INVITE
  136. Server: Asterisk PBX 11.23.0
  137. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  138. Supported: replaces, timer
  139. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fadc02a"
  140. Content-Length: 0
  141.  
  142.  
  143. <------------>
  144. Scheduling destruction of SIP dialog '1307273081-5060-21@BA.A.B.DE' in 6400 ms (Method: INVITE)
  145.  
  146. <--- SIP read from UDP:183.76.169.117:40408 --->
  147. ACK sip:09016192354@110.5.42.156 SIP/2.0
  148. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1348139891;rport
  149. From: <sip:200@110.5.42.156>;tag=158908947
  150. To: <sip:09016192354@110.5.42.156>;tag=as10029f86
  151. Call-ID: 1307273081-5060-21@BA.A.B.DE
  152. CSeq: 200 ACK
  153. Content-Length: 0
  154.  
  155. <------------->
  156. --- (7 headers 0 lines) ---
  157.  
  158. <--- SIP read from UDP:183.76.169.117:40408 --->
  159. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  160. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;rport
  161. From: <sip:200@110.5.42.156>;tag=158908947
  162. To: <sip:09016192354@110.5.42.156>
  163. Call-ID: 1307273081-5060-21@BA.A.B.DE
  164. CSeq: 201 INVITE
  165. Contact: <sip:200@10.0.1.34:5060>
  166. Authorization: Digest username="200", realm="asterisk", nonce="6fadc02a", uri="sip:09016192354@110.5.42.156", response="ead789077e536e449f40a53f96751a63", algorithm=MD5
  167. Max-Forwards: 70
  168. User-Agent: Grandstream GXP1625 1.0.2.27
  169. Privacy: none
  170. P-Preferred-Identity: <sip:200@110.5.42.156>
  171. Supported: replaces, path, timer
  172. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  173. Content-Type: application/sdp
  174. Accept: application/sdp, application/dtmf-relay
  175. Content-Length: 326
  176.  
  177. v=0
  178. o=200 8000 8000 IN IP4 10.0.1.34
  179. s=SIP Call
  180. c=IN IP4 10.0.1.34
  181. t=0 0
  182. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  183. a=sendrecv
  184. a=rtpmap:0 PCMU/8000
  185. a=ptime:20
  186. a=rtpmap:8 PCMA/8000
  187. a=rtpmap:18 G729/8000
  188. a=fmtp:18 annexb=no
  189. a=rtpmap:9 G722/8000
  190. a=rtpmap:2 G726-32/8000
  191. a=rtpmap:101 telephone-event/8000
  192. a=fmtp:101 0-15
  193. <------------->
  194. --- (17 headers 16 lines) ---
  195. Sending to 183.76.169.117:40408 (NAT)
  196. Using INVITE request as basis request - 1307273081-5060-21@BA.A.B.DE
  197. Found peer '200' for '200' from 183.76.169.117:40408
  198. == Using SIP RTP CoS mark 5
  199. Found RTP audio format 0
  200. Found RTP audio format 8
  201. Found RTP audio format 18
  202. Found RTP audio format 9
  203. Found RTP audio format 2
  204. Found RTP audio format 101
  205. Found audio description format PCMU for ID 0
  206. Found audio description format PCMA for ID 8
  207. Found audio description format G729 for ID 18
  208. Found audio description format G722 for ID 9
  209. Found audio description format G726-32 for ID 2
  210. Found audio description format telephone-event for ID 101
  211. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  212. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  213. Peer audio RTP is at port 10.0.1.34:5004
  214. Looking for 09016192354 in from-internal (domain 110.5.42.156)
  215. list_route: hop: <sip:200@10.0.1.34:5060>
  216.  
  217. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  218. SIP/2.0 100 Trying
  219. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;received=183.76.169.117;rport=40408
  220. From: <sip:200@110.5.42.156>;tag=158908947
  221. To: <sip:09016192354@110.5.42.156>
  222. Call-ID: 1307273081-5060-21@BA.A.B.DE
  223. CSeq: 201 INVITE
  224. Server: Asterisk PBX 11.23.0
  225. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  226. Supported: replaces, timer
  227. Session-Expires: 1800;refresher=uas
  228. Contact: <sip:09016192354@110.5.42.156:5060>
  229. Content-Length: 0
  230.  
  231.  
  232. <------------>
  233. -- Executing [09016192354@from-internal:1] ResetCDR("SIP/200-000002b6", "") in new stack
  234. -- Executing [09016192354@from-internal:2] NoCDR("SIP/200-000002b6", "") in new stack
  235. -- Executing [09016192354@from-internal:3] Progress("SIP/200-000002b6", "") in new stack
  236. Audio is at 18578
  237. Adding codec 100004 (alaw) to SDP
  238. Adding codec 100003 (ulaw) to SDP
  239. Adding non-codec 0x1 (telephone-event) to SDP
  240.  
  241. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  242. SIP/2.0 183 Session Progress
  243. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;received=183.76.169.117;rport=40408
  244. From: <sip:200@110.5.42.156>;tag=158908947
  245. To: <sip:09016192354@110.5.42.156>;tag=as55f3268e
  246. Call-ID: 1307273081-5060-21@BA.A.B.DE
  247. CSeq: 201 INVITE
  248. Server: Asterisk PBX 11.23.0
  249. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  250. Supported: replaces, timer
  251. Session-Expires: 1800;refresher=uas
  252. Contact: <sip:09016192354@110.5.42.156:5060>
  253. Content-Type: application/sdp
  254. Require: timer
  255. Content-Length: 260
  256.  
  257. v=0
  258. o=root 1008213452 1008213452 IN IP4 110.5.42.156
  259. s=Asterisk PBX 11.23.0
  260. c=IN IP4 110.5.42.156
  261. t=0 0
  262. m=audio 18578 RTP/AVP 8 0 101
  263. a=rtpmap:8 PCMA/8000
  264. a=rtpmap:0 PCMU/8000
  265. a=rtpmap:101 telephone-event/8000
  266. a=fmtp:101 0-16
  267. a=ptime:20
  268. a=sendrecv
  269.  
  270. <------------>
  271. -- Executing [09016192354@from-internal:4] Wait("SIP/200-000002b6", "1") in new stack
  272. > 0x7f2edc294e60 -- Probation passed - setting RTP source address to 183.76.169.117:46412
  273. > 0x7f2edc294e60 -- Probation passed - setting RTP source address to 183.76.169.117:46412
  274. -- Executing [09016192354@from-internal:5] Playback("SIP/200-000002b6", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
  275. -- <SIP/200-000002b6> Playing 'silence/1.alaw' (language 'en')
  276. -- <SIP/200-000002b6> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
  277. -- <SIP/200-000002b6> Playing 'check-number-dial-again.alaw' (language 'en')
  278.  
  279. <--- SIP read from UDP:183.76.169.117:40408 --->
  280. CANCEL sip:09016192354@110.5.42.156 SIP/2.0
  281. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;rport
  282. From: <sip:200@110.5.42.156>;tag=158908947
  283. To: <sip:09016192354@110.5.42.156>
  284. Call-ID: 1307273081-5060-21@BA.A.B.DE
  285. CSeq: 201 CANCEL
  286. Max-Forwards: 70
  287. User-Agent: Grandstream GXP1625 1.0.2.27
  288. Content-Length: 0
  289.  
  290. <------------->
  291. --- (9 headers 0 lines) ---
  292. Sending to 183.76.169.117:40408 (NAT)
  293.  
  294. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  295. SIP/2.0 487 Request Terminated
  296. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;received=183.76.169.117;rport=40408
  297. From: <sip:200@110.5.42.156>;tag=158908947
  298. To: <sip:09016192354@110.5.42.156>;tag=as55f3268e
  299. Call-ID: 1307273081-5060-21@BA.A.B.DE
  300. CSeq: 201 INVITE
  301. Server: Asterisk PBX 11.23.0
  302. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  303. Supported: replaces, timer
  304. Content-Length: 0
  305.  
  306.  
  307. <------------>
  308.  
  309. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  310. SIP/2.0 200 OK
  311. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;received=183.76.169.117;rport=40408
  312. From: <sip:200@110.5.42.156>;tag=158908947
  313. To: <sip:09016192354@110.5.42.156>;tag=as55f3268e
  314. Call-ID: 1307273081-5060-21@BA.A.B.DE
  315. CSeq: 201 CANCEL
  316. Server: Asterisk PBX 11.23.0
  317. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  318. Supported: replaces, timer
  319. Content-Length: 0
  320.  
  321.  
  322. <------------>
  323. == Spawn extension (from-internal, 09016192354, 5) exited non-zero on 'SIP/200-000002b6'
  324. -- Executing [h@from-internal:1] Macro("SIP/200-000002b6", "hangupcall") in new stack
  325. -- Executing [s@macro-hangupcall:1] ExecIf("SIP/200-000002b6", "0?Set(CDR(recordingfile)=.)") in new stack
  326. -- Executing [s@macro-hangupcall:2] GotoIf("SIP/200-000002b6", "1?theend") in new stack
  327. -- Goto (macro-hangupcall,s,4)
  328. -- Executing [s@macro-hangupcall:4] ExecIf("SIP/200-000002b6", "0?Set(CDR(recordingfile)=)") in new stack
  329. -- Executing [s@macro-hangupcall:5] Hangup("SIP/200-000002b6", "") in new stack
  330. == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/200-000002b6' in macro 'hangupcall'
  331. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-000002b6'
  332.  
  333. <--- SIP read from UDP:183.76.169.117:40408 --->
  334. ACK sip:09016192354@110.5.42.156 SIP/2.0
  335. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK2138054036;rport
  336. From: <sip:200@110.5.42.156>;tag=158908947
  337. To: <sip:09016192354@110.5.42.156>;tag=as55f3268e
  338. Call-ID: 1307273081-5060-21@BA.A.B.DE
  339. CSeq: 201 ACK
  340. Content-Length: 0
  341.  
  342. <------------->
  343. --- (7 headers 0 lines) ---
  344. Really destroying SIP dialog '1307273081-5060-21@BA.A.B.DE' Method: ACK
  345. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  346. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  347. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK496d8d74;rport
  348. Max-Forwards: 70
  349. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4bdd2f9a
  350. To: <sip:200@10.0.1.34:5060>
  351. Contact: <sip:asterisk@110.5.42.156:5060>
  352. Call-ID: 0627a23f4bf883967232921c03baed0c@110.5.42.156:5060
  353. CSeq: 102 OPTIONS
  354. User-Agent: Asterisk PBX 11.23.0
  355. Date: Tue, 23 Aug 2016 11:04:43 GMT
  356. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  357. Supported: replaces, timer
  358. Content-Length: 0
  359.  
  360.  
  361. ---
  362.  
  363. <--- SIP read from UDP:183.76.169.117:40408 --->
  364. SIP/2.0 200 OK
  365. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK496d8d74;rport=5060
  366. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as4bdd2f9a
  367. To: <sip:200@10.0.1.34:5060>;tag=1197011859
  368. Call-ID: 0627a23f4bf883967232921c03baed0c@110.5.42.156:5060
  369. CSeq: 102 OPTIONS
  370. Supported: replaces, path, timer
  371. User-Agent: Grandstream GXP1625 1.0.2.27
  372. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  373. Content-Length: 0
  374.  
  375. <------------->
  376. --- (10 headers 0 lines) ---
  377. Really destroying SIP dialog '0627a23f4bf883967232921c03baed0c@110.5.42.156:5060' Method: OPTIONS
  378. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  379. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  380. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK5e152ad7;rport
  381. Max-Forwards: 70
  382. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as58d66667
  383. To: <sip:200@10.0.1.34:5060>
  384. Contact: <sip:asterisk@110.5.42.156:5060>
  385. Call-ID: 30f157001657f3ae1bab1b0e077b6ed7@110.5.42.156:5060
  386. CSeq: 102 OPTIONS
  387. User-Agent: Asterisk PBX 11.23.0
  388. Date: Tue, 23 Aug 2016 11:05:03 GMT
  389. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  390. Supported: replaces, timer
  391. Content-Length: 0
  392.  
  393.  
  394. ---
  395.  
  396. <--- SIP read from UDP:183.76.169.117:40408 --->
  397. SIP/2.0 200 OK
  398. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK5e152ad7;rport=5060
  399. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as58d66667
  400. To: <sip:200@10.0.1.34:5060>;tag=916017021
  401. Call-ID: 30f157001657f3ae1bab1b0e077b6ed7@110.5.42.156:5060
  402. CSeq: 102 OPTIONS
  403. Supported: replaces, path, timer
  404. User-Agent: Grandstream GXP1625 1.0.2.27
  405. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  406. Content-Length: 0
  407.  
  408. <------------->
  409. --- (10 headers 0 lines) ---
  410. Really destroying SIP dialog '30f157001657f3ae1bab1b0e077b6ed7@110.5.42.156:5060' Method: OPTIONS
  411. localhost*CLI> exit
  412. Asterisk cleanly ending (0).
  413. Executing last minute cleanups
  414. [root@localhost ~]# asterisk -vrrr
  415. Asterisk 11.23.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  416. Created by Mark Spencer <markster@digium.com>
  417. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  418. This is free software, with components licensed under the GNU General Public
  419. License version 2 and other licenses; you are welcome to redistribute it under
  420. certain conditions. Type 'core show license' for details.
  421. =========================================================================
  422. Connected to Asterisk 11.23.0 currently running on localhost (pid = 13421)
  423. localhost*CLI> core set verbose 10
  424. Console verbose was 1 and is now 10.
  425. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  426. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  427. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK1aba4967;rport
  428. Max-Forwards: 70
  429. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as64497097
  430. To: <sip:200@10.0.1.34:5060>
  431. Contact: <sip:asterisk@110.5.42.156:5060>
  432. Call-ID: 402dbacb0c60d098497533bb6be32b12@110.5.42.156:5060
  433. CSeq: 102 OPTIONS
  434. User-Agent: Asterisk PBX 11.23.0
  435. Date: Tue, 23 Aug 2016 11:05:23 GMT
  436. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  437. Supported: replaces, timer
  438. Content-Length: 0
  439.  
  440.  
  441. ---
  442.  
  443. <--- SIP read from UDP:183.76.169.117:40408 --->
  444. SIP/2.0 200 OK
  445. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK1aba4967;rport=5060
  446. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as64497097
  447. To: <sip:200@10.0.1.34:5060>;tag=1595971486
  448. Call-ID: 402dbacb0c60d098497533bb6be32b12@110.5.42.156:5060
  449. CSeq: 102 OPTIONS
  450. Supported: replaces, path, timer
  451. User-Agent: Grandstream GXP1625 1.0.2.27
  452. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  453. Content-Length: 0
  454.  
  455. <------------->
  456. --- (10 headers 0 lines) ---
  457. Really destroying SIP dialog '402dbacb0c60d098497533bb6be32b12@110.5.42.156:5060' Method: OPTIONS
  458.  
  459. <--- SIP read from UDP:183.76.169.117:40408 --->
  460. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  461. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1568125717;rport
  462. From: <sip:200@110.5.42.156>;tag=725190583
  463. To: <sip:09016192354@110.5.42.156>
  464. Call-ID: 876093535-5060-22@BA.A.B.DE
  465. CSeq: 210 INVITE
  466. Contact: <sip:200@10.0.1.34:5060>
  467. Max-Forwards: 70
  468. User-Agent: Grandstream GXP1625 1.0.2.27
  469. Privacy: none
  470. P-Preferred-Identity: <sip:200@110.5.42.156>
  471. Supported: replaces, path, timer
  472. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  473. Content-Type: application/sdp
  474. Accept: application/sdp, application/dtmf-relay
  475. Content-Length: 326
  476.  
  477. v=0
  478. o=200 8000 8000 IN IP4 10.0.1.34
  479. s=SIP Call
  480. c=IN IP4 10.0.1.34
  481. t=0 0
  482. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  483. a=sendrecv
  484. a=rtpmap:0 PCMU/8000
  485. a=ptime:20
  486. a=rtpmap:8 PCMA/8000
  487. a=rtpmap:18 G729/8000
  488. a=fmtp:18 annexb=no
  489. a=rtpmap:9 G722/8000
  490. a=rtpmap:2 G726-32/8000
  491. a=rtpmap:101 telephone-event/8000
  492. a=fmtp:101 0-15
  493. <------------->
  494. --- (16 headers 16 lines) ---
  495. Sending to 183.76.169.117:40408 (NAT)
  496. Sending to 183.76.169.117:40408 (NAT)
  497. Using INVITE request as basis request - 876093535-5060-22@BA.A.B.DE
  498. Found peer '200' for '200' from 183.76.169.117:40408
  499.  
  500. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  501. SIP/2.0 401 Unauthorized
  502. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1568125717;received=183.76.169.117;rport=40408
  503. From: <sip:200@110.5.42.156>;tag=725190583
  504. To: <sip:09016192354@110.5.42.156>;tag=as5432357f
  505. Call-ID: 876093535-5060-22@BA.A.B.DE
  506. CSeq: 210 INVITE
  507. Server: Asterisk PBX 11.23.0
  508. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  509. Supported: replaces, timer
  510. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c155349"
  511. Content-Length: 0
  512.  
  513.  
  514. <------------>
  515. Scheduling destruction of SIP dialog '876093535-5060-22@BA.A.B.DE' in 6400 ms (Method: INVITE)
  516.  
  517. <--- SIP read from UDP:183.76.169.117:40408 --->
  518. ACK sip:09016192354@110.5.42.156 SIP/2.0
  519. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK1568125717;rport
  520. From: <sip:200@110.5.42.156>;tag=725190583
  521. To: <sip:09016192354@110.5.42.156>;tag=as5432357f
  522. Call-ID: 876093535-5060-22@BA.A.B.DE
  523. CSeq: 210 ACK
  524. Content-Length: 0
  525.  
  526. <------------->
  527. --- (7 headers 0 lines) ---
  528.  
  529. <--- SIP read from UDP:183.76.169.117:40408 --->
  530. INVITE sip:09016192354@110.5.42.156 SIP/2.0
  531. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK606829635;rport
  532. From: <sip:200@110.5.42.156>;tag=725190583
  533. To: <sip:09016192354@110.5.42.156>
  534. Call-ID: 876093535-5060-22@BA.A.B.DE
  535. CSeq: 211 INVITE
  536. Contact: <sip:200@10.0.1.34:5060>
  537. Authorization: Digest username="200", realm="asterisk", nonce="2c155349", uri="sip:09016192354@110.5.42.156", response="cbdbc59aa877b3f257b718cb507ab599", algorithm=MD5
  538. Max-Forwards: 70
  539. User-Agent: Grandstream GXP1625 1.0.2.27
  540. Privacy: none
  541. P-Preferred-Identity: <sip:200@110.5.42.156>
  542. Supported: replaces, path, timer
  543. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  544. Content-Type: application/sdp
  545. Accept: application/sdp, application/dtmf-relay
  546. Content-Length: 326
  547.  
  548. v=0
  549. o=200 8000 8000 IN IP4 10.0.1.34
  550. s=SIP Call
  551. c=IN IP4 10.0.1.34
  552. t=0 0
  553. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  554. a=sendrecv
  555. a=rtpmap:0 PCMU/8000
  556. a=ptime:20
  557. a=rtpmap:8 PCMA/8000
  558. a=rtpmap:18 G729/8000
  559. a=fmtp:18 annexb=no
  560. a=rtpmap:9 G722/8000
  561. a=rtpmap:2 G726-32/8000
  562. a=rtpmap:101 telephone-event/8000
  563. a=fmtp:101 0-15
  564. <------------->
  565. --- (17 headers 16 lines) ---
  566. Sending to 183.76.169.117:40408 (NAT)
  567. Using INVITE request as basis request - 876093535-5060-22@BA.A.B.DE
  568. Found peer '200' for '200' from 183.76.169.117:40408
  569. == Using SIP RTP CoS mark 5
  570. Found RTP audio format 0
  571. Found RTP audio format 8
  572. Found RTP audio format 18
  573. Found RTP audio format 9
  574. Found RTP audio format 2
  575. Found RTP audio format 101
  576. Found audio description format PCMU for ID 0
  577. Found audio description format PCMA for ID 8
  578. Found audio description format G729 for ID 18
  579. Found audio description format G722 for ID 9
  580. Found audio description format G726-32 for ID 2
  581. Found audio description format telephone-event for ID 101
  582. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  583. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  584. Peer audio RTP is at port 10.0.1.34:5004
  585. Looking for 09016192354 in from-internal (domain 110.5.42.156)
  586. list_route: hop: <sip:200@10.0.1.34:5060>
  587.  
  588. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  589. SIP/2.0 100 Trying
  590. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK606829635;received=183.76.169.117;rport=40408
  591. From: <sip:200@110.5.42.156>;tag=725190583
  592. To: <sip:09016192354@110.5.42.156>
  593. Call-ID: 876093535-5060-22@BA.A.B.DE
  594. CSeq: 211 INVITE
  595. Server: Asterisk PBX 11.23.0
  596. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  597. Supported: replaces, timer
  598. Session-Expires: 1800;refresher=uas
  599. Contact: <sip:09016192354@110.5.42.156:5060>
  600. Content-Length: 0
  601.  
  602.  
  603. <------------>
  604. -- Executing [09016192354@from-internal:1] ResetCDR("SIP/200-000002b7", "") in new stack
  605. -- Executing [09016192354@from-internal:2] NoCDR("SIP/200-000002b7", "") in new stack
  606. -- Executing [09016192354@from-internal:3] Progress("SIP/200-000002b7", "") in new stack
  607. Audio is at 12308
  608. Adding codec 100004 (alaw) to SDP
  609. Adding codec 100003 (ulaw) to SDP
  610. Adding non-codec 0x1 (telephone-event) to SDP
  611.  
  612. <--- Transmitting (NAT) to 183.76.169.117:40408 --->
  613. SIP/2.0 183 Session Progress
  614. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK606829635;received=183.76.169.117;rport=40408
  615. From: <sip:200@110.5.42.156>;tag=725190583
  616. To: <sip:09016192354@110.5.42.156>;tag=as29363394
  617. Call-ID: 876093535-5060-22@BA.A.B.DE
  618. CSeq: 211 INVITE
  619. Server: Asterisk PBX 11.23.0
  620. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  621. Supported: replaces, timer
  622. Session-Expires: 1800;refresher=uas
  623. Contact: <sip:09016192354@110.5.42.156:5060>
  624. Content-Type: application/sdp
  625. Require: timer
  626. Content-Length: 258
  627.  
  628. v=0
  629. o=root 778001177 778001177 IN IP4 110.5.42.156
  630. s=Asterisk PBX 11.23.0
  631. c=IN IP4 110.5.42.156
  632. t=0 0
  633. m=audio 12308 RTP/AVP 8 0 101
  634. a=rtpmap:8 PCMA/8000
  635. a=rtpmap:0 PCMU/8000
  636. a=rtpmap:101 telephone-event/8000
  637. a=fmtp:101 0-16
  638. a=ptime:20
  639. a=sendrecv
  640.  
  641. <------------>
  642. -- Executing [09016192354@from-internal:4] Wait("SIP/200-000002b7", "1") in new stack
  643. > 0x7f2edc2405b0 -- Probation passed - setting RTP source address to 183.76.169.117:46412
  644. > 0x7f2edc2405b0 -- Probation passed - setting RTP source address to 183.76.169.117:46412
  645. -- Executing [09016192354@from-internal:5] Playback("SIP/200-000002b7", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
  646. -- <SIP/200-000002b7> Playing 'silence/1.alaw' (language 'en')
  647. -- <SIP/200-000002b7> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
  648. -- <SIP/200-000002b7> Playing 'check-number-dial-again.alaw' (language 'en')
  649. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  650. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  651. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK7611653f;rport
  652. Max-Forwards: 70
  653. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as7a253a57
  654. To: <sip:200@10.0.1.34:5060>
  655. Contact: <sip:asterisk@110.5.42.156:5060>
  656. Call-ID: 484c97050535d6a465fbfe9019bcecf6@110.5.42.156:5060
  657. CSeq: 102 OPTIONS
  658. User-Agent: Asterisk PBX 11.23.0
  659. Date: Tue, 23 Aug 2016 11:05:43 GMT
  660. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  661. Supported: replaces, timer
  662. Content-Length: 0
  663.  
  664.  
  665. ---
  666.  
  667. <--- SIP read from UDP:183.76.169.117:40408 --->
  668. SIP/2.0 200 OK
  669. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK7611653f;rport=5060
  670. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as7a253a57
  671. To: <sip:200@10.0.1.34:5060>;tag=608638625
  672. Call-ID: 484c97050535d6a465fbfe9019bcecf6@110.5.42.156:5060
  673. CSeq: 102 OPTIONS
  674. Supported: replaces, path, timer
  675. User-Agent: Grandstream GXP1625 1.0.2.27
  676. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  677. Content-Length: 0
  678.  
  679. <------------->
  680. --- (10 headers 0 lines) ---
  681. Really destroying SIP dialog '484c97050535d6a465fbfe9019bcecf6@110.5.42.156:5060' Method: OPTIONS
  682. -- Executing [09016192354@from-internal:6] Wait("SIP/200-000002b7", "1") in new stack
  683. -- Executing [09016192354@from-internal:7] Congestion("SIP/200-000002b7", "20") in new stack
  684.  
  685. <--- Reliably Transmitting (NAT) to 183.76.169.117:40408 --->
  686. SIP/2.0 503 Service Unavailable
  687. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK606829635;received=183.76.169.117;rport=40408
  688. From: <sip:200@110.5.42.156>;tag=725190583
  689. To: <sip:09016192354@110.5.42.156>;tag=as29363394
  690. Call-ID: 876093535-5060-22@BA.A.B.DE
  691. CSeq: 211 INVITE
  692. Server: Asterisk PBX 11.23.0
  693. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  694. Supported: replaces, timer
  695. Session-Expires: 1800;refresher=uas
  696. Content-Length: 0
  697.  
  698.  
  699. <------------>
  700. [2016-08-23 20:05:44] WARNING[8814][C-000002c6]: channel.c:4861 ast_prod: Prodding channel 'SIP/200-000002b7' failed
  701. == Spawn extension (from-internal, 09016192354, 7) exited non-zero on 'SIP/200-000002b7'
  702. -- Executing [h@from-internal:1] Macro("SIP/200-000002b7", "hangupcall") in new stack
  703. -- Executing [s@macro-hangupcall:1] ExecIf("SIP/200-000002b7", "0?Set(CDR(recordingfile)=.)") in new stack
  704. -- Executing [s@macro-hangupcall:2] GotoIf("SIP/200-000002b7", "1?theend") in new stack
  705. -- Goto (macro-hangupcall,s,4)
  706. -- Executing [s@macro-hangupcall:4] ExecIf("SIP/200-000002b7", "0?Set(CDR(recordingfile)=)") in new stack
  707. -- Executing [s@macro-hangupcall:5] Hangup("SIP/200-000002b7", "") in new stack
  708. == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/200-000002b7' in macro 'hangupcall'
  709. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-000002b7'
  710.  
  711. <--- SIP read from UDP:183.76.169.117:40408 --->
  712. ACK sip:09016192354@110.5.42.156 SIP/2.0
  713. Via: SIP/2.0/UDP 10.0.1.34:5060;branch=z9hG4bK606829635;rport
  714. From: <sip:200@110.5.42.156>;tag=725190583
  715. To: <sip:09016192354@110.5.42.156>;tag=as29363394
  716. Call-ID: 876093535-5060-22@BA.A.B.DE
  717. CSeq: 211 ACK
  718. Content-Length: 0
  719.  
  720. <------------->
  721. --- (7 headers 0 lines) ---
  722. Really destroying SIP dialog '876093535-5060-22@BA.A.B.DE' Method: ACK
  723. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  724. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  725. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK62a0a243;rport
  726. Max-Forwards: 70
  727. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as11d4c26a
  728. To: <sip:200@10.0.1.34:5060>
  729. Contact: <sip:asterisk@110.5.42.156:5060>
  730. Call-ID: 76bb09c713aa25bb471d1ce42458e531@110.5.42.156:5060
  731. CSeq: 102 OPTIONS
  732. User-Agent: Asterisk PBX 11.23.0
  733. Date: Tue, 23 Aug 2016 11:06:03 GMT
  734. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  735. Supported: replaces, timer
  736. Content-Length: 0
  737.  
  738.  
  739. ---
  740.  
  741. <--- SIP read from UDP:183.76.169.117:40408 --->
  742. SIP/2.0 200 OK
  743. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK62a0a243;rport=5060
  744. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as11d4c26a
  745. To: <sip:200@10.0.1.34:5060>;tag=912879269
  746. Call-ID: 76bb09c713aa25bb471d1ce42458e531@110.5.42.156:5060
  747. CSeq: 102 OPTIONS
  748. Supported: replaces, path, timer
  749. User-Agent: Grandstream GXP1625 1.0.2.27
  750. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  751. Content-Length: 0
  752.  
  753. <------------->
  754. --- (10 headers 0 lines) ---
  755. Really destroying SIP dialog '76bb09c713aa25bb471d1ce42458e531@110.5.42.156:5060' Method: OPTIONS
  756. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  757. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  758. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK49972f81;rport
  759. Max-Forwards: 70
  760. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1f03fab8
  761. To: <sip:200@10.0.1.34:5060>
  762. Contact: <sip:asterisk@110.5.42.156:5060>
  763. Call-ID: 38dc50585da1b6403f249d937d5d84ed@110.5.42.156:5060
  764. CSeq: 102 OPTIONS
  765. User-Agent: Asterisk PBX 11.23.0
  766. Date: Tue, 23 Aug 2016 11:06:23 GMT
  767. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  768. Supported: replaces, timer
  769. Content-Length: 0
  770.  
  771.  
  772. ---
  773.  
  774. <--- SIP read from UDP:183.76.169.117:40408 --->
  775. SIP/2.0 200 OK
  776. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK49972f81;rport=5060
  777. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1f03fab8
  778. To: <sip:200@10.0.1.34:5060>;tag=1051142307
  779. Call-ID: 38dc50585da1b6403f249d937d5d84ed@110.5.42.156:5060
  780. CSeq: 102 OPTIONS
  781. Supported: replaces, path, timer
  782. User-Agent: Grandstream GXP1625 1.0.2.27
  783. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  784. Content-Length: 0
  785.  
  786. <------------->
  787. --- (10 headers 0 lines) ---
  788. Really destroying SIP dialog '38dc50585da1b6403f249d937d5d84ed@110.5.42.156:5060' Method: OPTIONS
  789. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  790. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  791. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK79bd040a;rport
  792. Max-Forwards: 70
  793. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1478ea01
  794. To: <sip:200@10.0.1.34:5060>
  795. Contact: <sip:asterisk@110.5.42.156:5060>
  796. Call-ID: 3775f64b133e09233843edac219c224f@110.5.42.156:5060
  797. CSeq: 102 OPTIONS
  798. User-Agent: Asterisk PBX 11.23.0
  799. Date: Tue, 23 Aug 2016 11:06:43 GMT
  800. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  801. Supported: replaces, timer
  802. Content-Length: 0
  803.  
  804.  
  805. ---
  806.  
  807. <--- SIP read from UDP:183.76.169.117:40408 --->
  808. SIP/2.0 200 OK
  809. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK79bd040a;rport=5060
  810. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as1478ea01
  811. To: <sip:200@10.0.1.34:5060>;tag=789447150
  812. Call-ID: 3775f64b133e09233843edac219c224f@110.5.42.156:5060
  813. CSeq: 102 OPTIONS
  814. Supported: replaces, path, timer
  815. User-Agent: Grandstream GXP1625 1.0.2.27
  816. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  817. Content-Length: 0
  818.  
  819. <------------->
  820. --- (10 headers 0 lines) ---
  821. Really destroying SIP dialog '3775f64b133e09233843edac219c224f@110.5.42.156:5060' Method: OPTIONS
  822. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  823. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  824. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK7c04f226;rport
  825. Max-Forwards: 70
  826. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as7ea96a71
  827. To: <sip:200@10.0.1.34:5060>
  828. Contact: <sip:asterisk@110.5.42.156:5060>
  829. Call-ID: 7237d4831b0d70f372a1e64e6094a8f1@110.5.42.156:5060
  830. CSeq: 102 OPTIONS
  831. User-Agent: Asterisk PBX 11.23.0
  832. Date: Tue, 23 Aug 2016 11:07:03 GMT
  833. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  834. Supported: replaces, timer
  835. Content-Length: 0
  836.  
  837.  
  838. ---
  839.  
  840. <--- SIP read from UDP:183.76.169.117:40408 --->
  841. SIP/2.0 200 OK
  842. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK7c04f226;rport=5060
  843. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as7ea96a71
  844. To: <sip:200@10.0.1.34:5060>;tag=1783571373
  845. Call-ID: 7237d4831b0d70f372a1e64e6094a8f1@110.5.42.156:5060
  846. CSeq: 102 OPTIONS
  847. Supported: replaces, path, timer
  848. User-Agent: Grandstream GXP1625 1.0.2.27
  849. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  850. Content-Length: 0
  851.  
  852. <------------->
  853. --- (10 headers 0 lines) ---
  854. Really destroying SIP dialog '7237d4831b0d70f372a1e64e6094a8f1@110.5.42.156:5060' Method: OPTIONS
  855. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  856. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  857. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK699e60c6;rport
  858. Max-Forwards: 70
  859. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5db22f33
  860. To: <sip:200@10.0.1.34:5060>
  861. Contact: <sip:asterisk@110.5.42.156:5060>
  862. Call-ID: 3e9077931962f254706227f805690197@110.5.42.156:5060
  863. CSeq: 102 OPTIONS
  864. User-Agent: Asterisk PBX 11.23.0
  865. Date: Tue, 23 Aug 2016 11:07:23 GMT
  866. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  867. Supported: replaces, timer
  868. Content-Length: 0
  869.  
  870.  
  871. ---
  872.  
  873. <--- SIP read from UDP:183.76.169.117:40408 --->
  874. SIP/2.0 200 OK
  875. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK699e60c6;rport=5060
  876. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as5db22f33
  877. To: <sip:200@10.0.1.34:5060>;tag=554829096
  878. Call-ID: 3e9077931962f254706227f805690197@110.5.42.156:5060
  879. CSeq: 102 OPTIONS
  880. Supported: replaces, path, timer
  881. User-Agent: Grandstream GXP1625 1.0.2.27
  882. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  883. Content-Length: 0
  884.  
  885. <------------->
  886. --- (10 headers 0 lines) ---
  887. Really destroying SIP dialog '3e9077931962f254706227f805690197@110.5.42.156:5060' Method: OPTIONS
  888. Reliably Transmitting (NAT) to 183.76.169.117:40408:
  889. OPTIONS sip:200@10.0.1.34:5060 SIP/2.0
  890. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK5b417ce2;rport
  891. Max-Forwards: 70
  892. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as71e4918d
  893. To: <sip:200@10.0.1.34:5060>
  894. Contact: <sip:asterisk@110.5.42.156:5060>
  895. Call-ID: 6c6ccd246fcef8bd06ce53505d77694f@110.5.42.156:5060
  896. CSeq: 102 OPTIONS
  897. User-Agent: Asterisk PBX 11.23.0
  898. Date: Tue, 23 Aug 2016 11:07:43 GMT
  899. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  900. Supported: replaces, timer
  901. Content-Length: 0
  902.  
  903.  
  904. ---
  905.  
  906. <--- SIP read from UDP:183.76.169.117:40408 --->
  907. SIP/2.0 200 OK
  908. Via: SIP/2.0/UDP 110.5.42.156:5060;branch=z9hG4bK5b417ce2;rport=5060
  909. From: "asterisk" <sip:asterisk@110.5.42.156>;tag=as71e4918d
  910. To: <sip:200@10.0.1.34:5060>;tag=1577425287
  911. Call-ID: 6c6ccd246fcef8bd06ce53505d77694f@110.5.42.156:5060
  912. CSeq: 102 OPTIONS
  913. Supported: replaces, path, timer
  914. User-Agent: Grandstream GXP1625 1.0.2.27
  915. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  916. Content-Length: 0
  917.  
  918. <------------->
  919. --- (10 headers 0 lines) ---
  920. Really destroying SIP dialog '6c6ccd246fcef8bd06ce53505d77694f@110.5.42.156:5060' Method: OPTIONS
  921. localhost*CLI>
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