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  1. CSeq: 27061 REGISTER
  2. User-Agent: CSipSimple r944 / su370-8
  3. Contact: <sip:androidsip01@192.168.0.206:59521;ob>
  4. Expires: 900
  5. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  6. Content-Length: 0
  7.  
  8. <------------->
  9. --- (13 headers 0 lines) ---
  10. Sending to 192.168.0.206:59521 (no NAT)
  11.  
  12. <--- Transmitting (no NAT) to 192.168.0.206:59521 --->
  13. SIP/2.0 401 Unauthorized
  14. Via: SIP/2.0/UDP 192.168.0.206:59521;branch=z9hG4bKPju.dotU58KnJYVqoIgLHrNBCAubSSNy8v;received=192.168.0.206;rport=59521
  15. From: <sip:androidsip01@192.168.0.236>;tag=DhSaEh4p3PZXk9Uf3T7.1PnZFCVO7y1q
  16. To: <sip:androidsip01@192.168.0.236>;tag=as7f862073
  17. Call-ID: eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk
  18. CSeq: 27061 REGISTER
  19. Server: Asterisk PBX 1.8.4.4
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  21. Supported: replaces, timer
  22. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a1e9f47"
  23. Content-Length: 0
  24.  
  25.  
  26. <------------>
  27. Scheduling destruction of SIP dialog 'eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk' in 32000 ms (Method: REGISTER)
  28.  
  29. <--- SIP read from UDP:192.168.0.206:59521 --->
  30. REGISTER sip:192.168.0.236 SIP/2.0
  31. Via: SIP/2.0/UDP 192.168.0.206:59521;rport;branch=z9hG4bKPjD3TPcJZOjkQcYRPrXT9X.Ro-0iRJcq.H
  32. Route: <sip:192.168.0.236;transport=udp;lr>
  33. Max-Forwards: 70
  34. From: <sip:androidsip01@192.168.0.236>;tag=DhSaEh4p3PZXk9Uf3T7.1PnZFCVO7y1q
  35. To: <sip:androidsip01@192.168.0.236>
  36. Call-ID: eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk
  37. CSeq: 27062 REGISTER
  38. User-Agent: CSipSimple r944 / su370-8
  39. Contact: <sip:androidsip01@192.168.0.206:59521;ob>
  40. Expires: 900
  41. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  42. Authorization: Digest username="androidsip01", realm="asterisk", nonce="4a1e9f47", uri="sip:192.168.0.236", response="2e61e08729798f4b048d67e09f5c16b5", algorithm=MD5
  43. Content-Length: 0
  44.  
  45. <------------->
  46. --- (14 headers 0 lines) ---
  47. Sending to 192.168.0.206:59521 (no NAT)
  48. -- Registered SIP 'androidsip01' at 192.168.0.206:59521
  49.  
  50. <--- Transmitting (no NAT) to 192.168.0.206:59521 --->
  51. SIP/2.0 200 OK
  52. Via: SIP/2.0/UDP 192.168.0.206:59521;branch=z9hG4bKPjD3TPcJZOjkQcYRPrXT9X.Ro-0iRJcq.H;received=192.168.0.206;rport=59521
  53. From: <sip:androidsip01@192.168.0.236>;tag=DhSaEh4p3PZXk9Uf3T7.1PnZFCVO7y1q
  54. To: <sip:androidsip01@192.168.0.236>;tag=as7f862073
  55. Call-ID: eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk
  56. CSeq: 27062 REGISTER
  57. Server: Asterisk PBX 1.8.4.4
  58. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  59. Supported: replaces, timer
  60. Expires: 900
  61. Contact: <sip:androidsip01@192.168.0.206:59521;ob>;expires=900
  62. Date: Thu, 04 Aug 2011 23:23:56 GMT
  63. Content-Length: 0
  64.  
  65.  
  66. <------------>
  67. Scheduling destruction of SIP dialog 'eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk' in 32000 ms (Method: REGISTER)
  68. Really destroying SIP dialog 'eBKc3dc.tP-9uMlSY0I7OnmRMcOTivzk' Method: REGISTER
  69. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  70. OPTIONS sip:sip.flowroute.com SIP/2.0
  71. Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK4ffc12ed;rport
  72. Max-Forwards: 70
  73. From: "asterisk" <sip:asterisk@192.168.0.236>;tag=as13fd4229
  74. To: <sip:sip.flowroute.com>
  75. Contact: <sip:asterisk@192.168.0.236:5060>
  76. Call-ID: 3f8088db0c2daaaa3b927b210b014506@192.168.0.236:5060
  77. CSeq: 102 OPTIONS
  78. User-Agent: Asterisk PBX 1.8.4.4
  79. Date: Thu, 04 Aug 2011 23:24:31 GMT
  80. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  81. Supported: replaces, timer
  82. Content-Length: 0
  83.  
  84.  
  85. ---
  86.  
  87. <--- SIP read from UDP:216.115.69.144:5060 --->
  88. SIP/2.0 200 OK
  89. Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK4ffc12ed;rport=5060;received=121.135.82.142
  90. From: "asterisk" <sip:asterisk@192.168.0.236>;tag=as13fd4229
  91. To: <sip:sip.flowroute.com>;tag=20e1698a3241cbcc6677a39fcb65a0aa.52d1
  92. Call-ID: 3f8088db0c2daaaa3b927b210b014506@192.168.0.236:5060
  93. CSeq: 102 OPTIONS
  94. Accept: */*
  95. Accept-Encoding:
  96. Accept-Language: en
  97. Supported:
  98. Content-Length: 0
  99.  
  100. <------------->
  101. --- (11 headers 0 lines) ---
  102. Really destroying SIP dialog '3f8088db0c2daaaa3b927b210b014506@192.168.0.236:5060' Method: OPTIONS
  103. asterisknow*CLI> sip set debug off
  104. SIP Debugging Disabled
  105. asterisknow*CLI>
  106. [Aug 5 08:33:55] NOTICE[2561]: chan_sip.c:23613 handle_request_subscribe: Received SIP subscribe for peer without mailbox: david_sip
  107. == Using SIP RTP CoS mark 5
  108. -- Executing [12012156850@from-flowroute:1] Answer("SIP/flowroute-0000002f", "") in new stack
  109. -- Auto fallthrough, channel 'SIP/flowroute-0000002f' status is 'UNKNOWN'
  110. asterisknow*CLI> core set verbose 3
  111. Verbosity is at least 3
  112. -- Registered SIP 'androidsip01' at 192.168.0.206:47326
  113. == Using SIP RTP CoS mark 5
  114. -- Executing [12012156850@from-flowroute:1] Answer("SIP/flowroute-00000030", "") in new stack
  115. -- Auto fallthrough, channel 'SIP/flowroute-00000030' status is 'UNKNOWN'
  116. asterisknow*CLI> sip debug on
  117. No such command 'sip debug on' (type 'core show help sip debug on' for other possible commands)
  118. asterisknow*CLI> sip set debug on
  119. SIP Debugging enabled
  120. asterisknow*CLI>
  121.  
  122.  
  123.  
  124.  
  125.  
  126.  
  127.  
  128.  
  129.  
  130.  
  131.  
  132.  
  133.  
  134.  
  135.  
  136.  
  137.  
  138.  
  139.  
  140.  
  141.  
  142.  
  143.  
  144.  
  145.  
  146.  
  147.  
  148.  
  149.  
  150.  
  151.  
  152.  
  153.  
  154.  
  155.  
  156.  
  157.  
  158.  
  159.  
  160.  
  161. <--- SIP read from UDP:216.115.69.144:5060 --->
  162. INVITE sip:12012156850@192.168.0.236:5060 SIP/2.0
  163. Record-Route: <sip:216.115.69.144;lr>
  164. Record-Route: <sip:216.115.69.132;lr>
  165. To: <sip:+12012156850@flowroute.com>
  166. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  167. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0
  168. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  169. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  170. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  171. Call-ID: 822948119_39439356@4.55.17.35
  172. CSeq: 30671 INVITE
  173. Max-Forwards: 63
  174. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
  175. Contact: "david.juhl" <sip:Anonymous@4.55.17.35:5060>
  176. Content-Length: 225
  177. Content-Type: application/sdp
  178. P-Asserted-Identity: "david.juhl " <sip:UNAVAILABLE@flowroute.com>
  179.  
  180. v=0
  181. o=- 3012 22342 IN IP4 4.55.17.2
  182. s=-
  183. c=IN IP4 4.55.17.2
  184. t=0 0
  185. m=audio 11406 RTP/AVP 0 18 101
  186. a=rtpmap:18 G729/8000
  187. a=fmtp:18 annexb=no
  188. a=rtpmap:101 telephone-event/8000
  189. a=fmtp:101 0-15
  190. a=ptime:20
  191. a=maxptime:20
  192. <------------->
  193. --- (17 headers 12 lines) ---
  194. Sending to 216.115.69.144:5060 (no NAT)
  195. Using INVITE request as basis request - 822948119_39439356@4.55.17.35
  196. Found peer 'flowroute' for 'UNAVAILABLE' from 216.115.69.144:5060
  197. == Using SIP RTP CoS mark 5
  198. Found RTP audio format 0
  199. Found RTP audio format 18
  200. Found RTP audio format 101
  201. Found audio description format G729 for ID 18
  202. Found audio description format telephone-event for ID 101
  203. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  204. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  205. Peer audio RTP is at port 4.55.17.2:11406
  206. Looking for 12012156850 in from-flowroute (domain 192.168.0.236:5060)
  207. list_route: hop: <sip:216.115.69.144;lr>
  208. list_route: hop: <sip:216.115.69.132;lr>
  209.  
  210. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  211. SIP/2.0 100 Trying
  212. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0;received=216.115.69.144;rport=5060
  213. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  214. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  215. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  216. Record-Route: <sip:216.115.69.144;lr>
  217. Record-Route: <sip:216.115.69.132;lr>
  218. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  219. To: <sip:+12012156850@flowroute.com>
  220. Call-ID: 822948119_39439356@4.55.17.35
  221. CSeq: 30671 INVITE
  222. Server: Asterisk PBX 1.8.4.4
  223. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  224. Supported: replaces, timer
  225. Contact: <sip:12012156850@192.168.0.236:5060>
  226. Content-Length: 0
  227.  
  228.  
  229. <------------>
  230. -- Executing [12012156850@from-flowroute:1] Answer("SIP/flowroute-00000031", "") in new stack
  231. Audio is at 5060
  232. Adding codec 0x4 (ulaw) to SDP
  233. Adding non-codec 0x1 (telephone-event) to SDP
  234.  
  235. <--- Reliably Transmitting (NAT) to 216.115.69.144:5060 --->
  236. SIP/2.0 200 OK
  237. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0;received=216.115.69.144;rport=5060
  238. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  239. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  240. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  241. Record-Route: <sip:216.115.69.144;lr>
  242. Record-Route: <sip:216.115.69.132;lr>
  243. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  244. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  245. Call-ID: 822948119_39439356@4.55.17.35
  246. CSeq: 30671 INVITE
  247. Server: Asterisk PBX 1.8.4.4
  248. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  249. Supported: replaces, timer
  250. Contact: <sip:12012156850@192.168.0.236:5060>
  251. Content-Type: application/sdp
  252. Content-Length: 236
  253.  
  254. v=0
  255. o=root 862549808 862549808 IN IP4 192.168.0.236
  256. s=Asterisk PBX 1.8.4.4
  257. c=IN IP4 192.168.0.236
  258. t=0 0
  259. m=audio 14898 RTP/AVP 0 101
  260. a=rtpmap:0 PCMU/8000
  261. a=rtpmap:101 telephone-event/8000
  262. a=fmtp:101 0-16
  263. a=ptime:20
  264. a=sendrecv
  265.  
  266. <------------>
  267.  
  268. <--- SIP read from UDP:216.115.69.144:5060 --->
  269. INVITE sip:12012156850@192.168.0.236:5060 SIP/2.0
  270. Record-Route: <sip:216.115.69.144;lr>
  271. Record-Route: <sip:216.115.69.132;lr>
  272. To: <sip:+12012156850@flowroute.com>
  273. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  274. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0
  275. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  276. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  277. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  278. Call-ID: 822948119_39439356@4.55.17.35
  279. CSeq: 30671 INVITE
  280. Max-Forwards: 63
  281. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
  282. Contact: "david.juhl" <sip:Anonymous@4.55.17.35:5060>
  283. Content-Length: 225
  284. Content-Type: application/sdp
  285. P-Asserted-Identity: "david.juhl " <sip:UNAVAILABLE@flowroute.com>
  286.  
  287. v=0
  288. o=- 3012 22342 IN IP4 4.55.17.2
  289. s=-
  290. c=IN IP4 4.55.17.2
  291. t=0 0
  292. m=audio 11406 RTP/AVP 0 18 101
  293. a=rtpmap:18 G729/8000
  294. a=fmtp:18 annexb=no
  295. a=rtpmap:101 telephone-event/8000
  296. a=fmtp:101 0-15
  297. a=ptime:20
  298. a=maxptime:20
  299. <------------->
  300. --- (17 headers 12 lines) ---
  301. Ignoring this INVITE request
  302.  
  303. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  304. SIP/2.0 100 Trying
  305. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0;received=216.115.69.144;rport=5060
  306. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  307. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  308. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  309. Record-Route: <sip:216.115.69.144;lr>
  310. Record-Route: <sip:216.115.69.132;lr>
  311. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  312. To: <sip:+12012156850@flowroute.com>
  313. Call-ID: 822948119_39439356@4.55.17.35
  314. CSeq: 30671 INVITE
  315. Server: Asterisk PBX 1.8.4.4
  316. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  317. Supported: replaces, timer
  318. Contact: <sip:12012156850@192.168.0.236:5060>
  319. Content-Length: 0
  320.  
  321.  
  322. <------------>
  323. Audio is at 5060
  324. Adding codec 0x4 (ulaw) to SDP
  325. Adding non-codec 0x1 (telephone-event) to SDP
  326.  
  327. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  328. SIP/2.0 200 OK
  329. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0;received=216.115.69.144;rport=5060
  330. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  331. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  332. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  333. Record-Route: <sip:216.115.69.144;lr>
  334. Record-Route: <sip:216.115.69.132;lr>
  335. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  336. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  337. Call-ID: 822948119_39439356@4.55.17.35
  338. CSeq: 30671 INVITE
  339. Server: Asterisk PBX 1.8.4.4
  340. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  341. Supported: replaces, timer
  342. Contact: <sip:12012156850@192.168.0.236:5060>
  343. Content-Type: application/sdp
  344. Content-Length: 236
  345.  
  346. v=0
  347. o=root 862549808 862549809 IN IP4 192.168.0.236
  348. s=Asterisk PBX 1.8.4.4
  349. c=IN IP4 192.168.0.236
  350. t=0 0
  351. m=audio 14898 RTP/AVP 0 101
  352. a=rtpmap:0 PCMU/8000
  353. a=rtpmap:101 telephone-event/8000
  354. a=fmtp:101 0-16
  355. a=ptime:20
  356. a=sendrecv
  357.  
  358. <------------>
  359. Retransmitting #1 (NAT) to 216.115.69.144:5060:
  360. SIP/2.0 200 OK
  361. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.aa9e6336e020b3e7801606f1050845da.0;received=216.115.69.144;rport=5060
  362. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK1035.21998b82058d015d073c3e6c89c2836f.0
  363. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.0916a7943d4c97a68bd070500f495afa.0
  364. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfd750c1df43382f
  365. Record-Route: <sip:216.115.69.144;lr>
  366. Record-Route: <sip:216.115.69.132;lr>
  367. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  368. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  369. Call-ID: 822948119_39439356@4.55.17.35
  370. CSeq: 30671 INVITE
  371. Server: Asterisk PBX 1.8.4.4
  372. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  373. Supported: replaces, timer
  374. Contact: <sip:12012156850@192.168.0.236:5060>
  375. Content-Type: application/sdp
  376. Content-Length: 236
  377.  
  378. v=0
  379. o=root 862549808 862549808 IN IP4 192.168.0.236
  380. s=Asterisk PBX 1.8.4.4
  381. c=IN IP4 192.168.0.236
  382. t=0 0
  383. m=audio 14898 RTP/AVP 0 101
  384. a=rtpmap:0 PCMU/8000
  385. a=rtpmap:101 telephone-event/8000
  386. a=fmtp:101 0-16
  387. a=ptime:20
  388. a=sendrecv
  389.  
  390. ---
  391. -- Auto fallthrough, channel 'SIP/flowroute-00000031' status is 'UNKNOWN'
  392. Scheduling destruction of SIP dialog '822948119_39439356@4.55.17.35' in 31296 ms (Method: INVITE)
  393.  
  394. <--- SIP read from UDP:216.115.69.144:5060 --->
  395. ACK sip:12012156850@121.135.82.142:5060 SIP/2.0
  396. Record-Route: <sip:216.115.69.144;lr>
  397. Record-Route: <sip:216.115.69.132;lr>
  398. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  399. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  400. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.990fac34b068707785675896fb864f34.0
  401. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.d8ef2a57ccd6ca93f1ed6f159d1ad4c4.0
  402. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfdd564f38d2b73f
  403. Call-ID: 822948119_39439356@4.55.17.35
  404. CSeq: 30671 ACK
  405. Max-Forwards: 68
  406. Content-Length: 0
  407.  
  408. <------------->
  409. --- (12 headers 0 lines) ---
  410. set_destination: Parsing <sip:216.115.69.144;lr> for address/port to send to
  411. set_destination: set destination to 216.115.69.144:5060
  412. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  413. BYE sip:Anonymous@4.55.17.35:5060 SIP/2.0
  414. Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK775ee603;rport
  415. Route: <sip:216.115.69.144;lr>,<sip:216.115.69.132;lr>
  416. Max-Forwards: 70
  417. From: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  418. To: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  419. Call-ID: 822948119_39439356@4.55.17.35
  420. CSeq: 102 BYE
  421. User-Agent: Asterisk PBX 1.8.4.4
  422. X-Asterisk-HangupCause: Unknown
  423. X-Asterisk-HangupCauseCode: 0
  424. Content-Length: 0
  425.  
  426.  
  427. ---
  428. Scheduling destruction of SIP dialog '822948119_39439356@4.55.17.35' in 31296 ms (Method: ACK)
  429.  
  430. <--- SIP read from UDP:216.115.69.144:5060 --->
  431. ACK sip:12012156850@121.135.82.142:5060 SIP/2.0
  432. Record-Route: <sip:216.115.69.144;lr>
  433. Record-Route: <sip:216.115.69.132;lr>
  434. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  435. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  436. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.c79c587f81086abea02aaa8e577fb824.0
  437. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.4c484900b1d42c2ee40e21233d9b5aa4.0
  438. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfde6c8b38d2b73f
  439. Call-ID: 822948119_39439356@4.55.17.35
  440. CSeq: 30671 ACK
  441. Max-Forwards: 68
  442. Content-Length: 0
  443.  
  444. <------------->
  445. --- (12 headers 0 lines) ---
  446.  
  447. <--- SIP read from UDP:216.115.69.144:5060 --->
  448. ACK sip:12012156850@121.135.82.142:5060 SIP/2.0
  449. Record-Route: <sip:216.115.69.144;lr>
  450. Record-Route: <sip:216.115.69.132;lr>
  451. From: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  452. To: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  453. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1035.6c8e7a1da1109bb827989b3e99c17255.0
  454. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK1035.7d64f9332ba405a6c77914acdde5fdbe.0
  455. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBbfdf344d38d2b73f
  456. Call-ID: 822948119_39439356@4.55.17.35
  457. CSeq: 30671 ACK
  458. Max-Forwards: 68
  459. Content-Length: 0
  460.  
  461. <------------->
  462. --- (12 headers 0 lines) ---
  463. Retransmitting #1 (NAT) to 216.115.69.144:5060:
  464. BYE sip:Anonymous@4.55.17.35:5060 SIP/2.0
  465. Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK775ee603;rport
  466. Route: <sip:216.115.69.144;lr>,<sip:216.115.69.132;lr>
  467. Max-Forwards: 70
  468. From: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  469. To: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  470. Call-ID: 822948119_39439356@4.55.17.35
  471. CSeq: 102 BYE
  472. User-Agent: Asterisk PBX 1.8.4.4
  473. X-Asterisk-HangupCause: Unknown
  474. X-Asterisk-HangupCauseCode: 0
  475. Content-Length: 0
  476.  
  477.  
  478. ---
  479.  
  480. <--- SIP read from UDP:216.115.69.144:5060 --->
  481. SIP/2.0 200 OK
  482. From: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  483. To: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  484. Via: SIP/2.0/UDP 192.168.0.236:5060;received=121.135.82.142;branch=z9hG4bK775ee603;rport=5060
  485. Call-ID: 822948119_39439356@4.55.17.35
  486. CSeq: 102 BYE
  487. Record-Route: <sip:216.115.69.132:5060;lr>
  488. Record-Route: <sip:216.115.69.144:5060;lr>
  489. Content-Length: 0
  490.  
  491. <------------->
  492. --- (9 headers 0 lines) ---
  493. SIP Response message for INCOMING dialog BYE arrived
  494. Really destroying SIP dialog '822948119_39439356@4.55.17.35' Method: ACK
  495.  
  496. <--- SIP read from UDP:216.115.69.144:5060 --->
  497. SIP/2.0 200 OK
  498. From: <sip:+12012156850@flowroute.com>;tag=as34fe54cf
  499. To: "david.juhl " <sip:UNAVAILABLE@flowroute.com>;tag=gK0d6390ff
  500. Via: SIP/2.0/UDP 192.168.0.236:5060;received=121.135.82.142;branch=z9hG4bK775ee603;rport=5060
  501. Call-ID: 822948119_39439356@4.55.17.35
  502. CSeq: 102 BYE
  503. Record-Route: <sip:216.115.69.132:5060;lr>
  504. Record-Route: <sip:216.115.69.144:5060;lr>
  505. Content-Length: 0
  506.  
  507. <------------->
  508. --- (9 headers 0 lines) ---
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