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  1. [Jul 17 12:19:30] VERBOSE[2534] chan_sip.c:
  2. <--- SIP read from UDP:192.168.80.131:5060 --->
  3. SUBSCRIBE sip:192.168.80.1 SIP/2.0
  4. Via: SIP/2.0/UDP 192.168.80.131:5060;branch=z9hG4bK-52cc7d
  5. From: <sip:202@192.168.80.1>;tag=72bc1badf5f894f1
  6. To: <sip:202@192.168.80.1>;tag=as27845af1
  7. Call-ID: 9f1cff5d-62cd8501@192.168.80.131
  8. CSeq: 9102 SUBSCRIBE
  9. Max-Forwards: 70
  10. Authorization: Digest username="202",realm="asterisk",nonce="303d56f0",uri="sip:192.168.80.1",algorithm=MD5,response="5369784c6c1293c0e8c1d2d0032b3cf0"
  11. Contact: <sip:202@192.168.80.131:5060>
  12. Expires: 2147483647
  13. Event: message-summary
  14. User-Agent: RTP300-3.1.22
  15. Content-Length: 0
  16.  
  17. <------------->
  18. [Jul 17 12:19:30] VERBOSE[2534] chan_sip.c: --- (13 headers 0 lines) ---
  19. [Jul 17 12:19:30] VERBOSE[2534] chan_sip.c:
  20. <--- Transmitting (NAT) to 192.168.80.131:5060 --->
  21. SIP/2.0 481 Call/Transaction Does Not Exist
  22. Via: SIP/2.0/UDP 192.168.80.131:5060;branch=z9hG4bK-52cc7d;received=192.168.80.131;rport=5060
  23. From: <sip:202@192.168.80.1>;tag=72bc1badf5f894f1
  24. To: <sip:202@192.168.80.1>;tag=as27845af1
  25. Call-ID: 9f1cff5d-62cd8501@192.168.80.131
  26. CSeq: 9102 SUBSCRIBE
  27. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  28. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  29. Supported: replaces, timer
  30. Content-Length: 0
  31.  
  32.  
  33. <------------>
  34. [Jul 17 12:19:30] VERBOSE[2534] chan_sip.c: Scheduling destruction of SIP dialog '9f1cff5d-62cd8501@192.168.80.131' in 32000 ms (Method: SUBSCRIBE)
  35. [Jul 17 12:19:31] VERBOSE[2534] chan_sip.c:
  36. <--- SIP read from UDP:192.168.80.1:5061 --->
  37. jaK
  38. <------------->
  39. [Jul 17 12:19:32] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog '8bf77291-9fd87fa4@192.168.80.131' Method: SUBSCRIBE
  40. [Jul 17 12:19:32] VERBOSE[2534] chan_sip.c:
  41. <--- SIP read from UDP:192.168.80.131:5060 --->
  42. SUBSCRIBE sip:192.168.80.1 SIP/2.0
  43. Via: SIP/2.0/UDP 192.168.80.131:5060;branch=z9hG4bK-4dca668a
  44. From: <sip:201@192.168.80.1>;tag=4108180ed1c15cd9
  45. To: <sip:201@192.168.80.1>;tag=as3a3675be
  46. Call-ID: 315ffbf2-b1e62a45@192.168.80.131
  47. CSeq: 6967 SUBSCRIBE
  48. Max-Forwards: 70
  49. Authorization: Digest username="201",realm="asterisk",nonce="759b6734",uri="sip:192.168.80.1",algorithm=MD5,response="82179f365bedf830e5458e046e1c7bc0"
  50. Contact: <sip:201@192.168.80.131:5060>
  51. Expires: 2147483647
  52. Event: message-summary
  53. User-Agent: RTP300-3.1.22
  54. Content-Length: 0
  55.  
  56. <------------->
  57. [Jul 17 12:19:32] VERBOSE[2534] chan_sip.c: --- (13 headers 0 lines) ---
  58. [Jul 17 12:19:32] VERBOSE[2534] chan_sip.c:
  59. <--- Transmitting (NAT) to 192.168.80.131:5060 --->
  60. SIP/2.0 481 Call/Transaction Does Not Exist
  61. Via: SIP/2.0/UDP 192.168.80.131:5060;branch=z9hG4bK-4dca668a;received=192.168.80.131;rport=5060
  62. From: <sip:201@192.168.80.1>;tag=4108180ed1c15cd9
  63. To: <sip:201@192.168.80.1>;tag=as3a3675be
  64. Call-ID: 315ffbf2-b1e62a45@192.168.80.131
  65. CSeq: 6967 SUBSCRIBE
  66. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  67. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  68. Supported: replaces, timer
  69. Content-Length: 0
  70.  
  71.  
  72. <------------>
  73. [Jul 17 12:19:32] VERBOSE[2534] chan_sip.c: Scheduling destruction of SIP dialog '315ffbf2-b1e62a45@192.168.80.131' in 32000 ms (Method: SUBSCRIBE)
  74. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog 'e6760161-126bfe9@192.168.80.131' Method: SUBSCRIBE
  75. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c:
  76. <--- SIP read from UDP:208.73.146.95:5060 --->
  77. INVITE sip:voip number@173.167.212.105:5060 SIP/2.0
  78. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1
  79. Call-Id: pcst1342541974696254284310@192.168.201.113
  80. Contact: <sip+1 lane line number@208.73.146.95:5060;transport=udp>
  81. Content-Length: 288
  82. Content-Type: application/sdp
  83. CSeq: 1 INVITE
  84. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  85. Session-Expires: 3600;refresher=uas
  86. Supported: timer
  87. To: <sip:voip number@192.168.101.100:5060>
  88. Max-Forwards: 70
  89.  
  90. v=0
  91. o=msc3 727 19257 IN IP4 208.73.146.95
  92. s=sip call
  93. c=IN IP4 208.73.146.95
  94. t=0 0
  95. m=audio 24526 RTP/AVP 0 8 18 101
  96. a=rtpmap:0 PCMU/8000
  97. a=rtpmap:8 PCMA/8000
  98. a=rtpmap:18 G729/8000
  99. a=fmtp:18 annexb=no
  100. a=rtpmap:101 telephone-event/8000
  101. a=fmtp:101 0-15
  102. a=sendrecv
  103. a=maxptime:20
  104. <------------->
  105. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: --- (12 headers 14 lines) ---
  106. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Sending to 208.73.146.95:5060 (NAT)
  107. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Using INVITE request as basis request - pcst1342541974696254284310@192.168.201.113
  108. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found peer 'voip number' for '+12153681640' from 208.73.146.95:5060
  109. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found RTP audio format 0
  110. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found RTP audio format 8
  111. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found RTP audio format 18
  112. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found RTP audio format 101
  113. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found audio description format PCMU for ID 0
  114. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found audio description format PCMA for ID 8
  115. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found audio description format G729 for ID 18
  116. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Found audio description format telephone-event for ID 101
  117. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  118. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  119. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Peer audio RTP is at port 208.73.146.95:24526
  120. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: Looking for voip number in from-trunk (domain 173.167.212.105)
  121. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c: list_route: hop: <sip+1 lane line number@208.73.146.95:5060;transport=udp>
  122. [Jul 17 12:19:34] VERBOSE[2534] chan_sip.c:
  123. <--- Transmitting (NAT) to 208.73.146.95:5060 --->
  124. SIP/2.0 100 Trying
  125. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1;received=208.73.146.95;rport=5060
  126. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  127. To: <sip:voip number@192.168.101.100:5060>
  128. Call-ID: pcst1342541974696254284310@192.168.201.113
  129. CSeq: 1 INVITE
  130. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  131. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  132. Supported: replaces, timer
  133. Session-Expires: 1800;refresher=uas
  134. Contact: <sip:voip number@173.167.212.105:5060>
  135. Content-Length: 0
  136.  
  137.  
  138. <------------>
  139. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Scheduling destruction of SIP dialog '511dc2c04392a4353f073d62418fd7d2@192.168.80.1:5060' in 32000 ms (Method: NOTIFY)
  140. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.1:5061:
  141. NOTIFY sip:200@192.168.80.1:5061;line=70cc019d1600592 SIP/2.0
  142. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5ffffbef;rport
  143. Max-Forwards: 70
  144. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as15101063
  145. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>
  146. Contact: <sip:asterisk@192.168.80.1:5060>
  147. Call-ID: 511dc2c04392a4353f073d62418fd7d2@192.168.80.1:5060
  148. CSeq: 102 NOTIFY
  149. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  150. Event: message-summary
  151. Content-Type: application/simple-message-summary
  152. Content-Length: 92
  153.  
  154. Messages-Waiting: no
  155. Message-Account: sip:asterisk@192.168.80.1
  156. Voice-Message: 0/0 (0/0)
  157.  
  158. ---
  159. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  160. <--- SIP read from UDP:192.168.80.1:5061 --->
  161. SIP/2.0 481 Subscription Does Not Exist
  162. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5ffffbef;rport=5060
  163. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as15101063
  164. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=1773849108
  165. Call-ID: 511dc2c04392a4353f073d62418fd7d2@192.168.80.1:5060
  166. CSeq: 102 NOTIFY
  167. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  168. Content-Length: 0
  169.  
  170. <------------->
  171. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  172. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog '511dc2c04392a4353f073d62418fd7d2@192.168.80.1:5060' Method: NOTIFY
  173. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Audio is at 15220
  174. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  175. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  176. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  177. [Jul 17 12:19:35] VERBOSE[24763] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.1:5061:
  178. INVITE sip:200@192.168.80.1:5061;line=70cc019d1600592 SIP/2.0
  179. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport
  180. Max-Forwards: 70
  181. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  182. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>
  183. Contact: <sip:voip number@192.168.80.1:5060>
  184. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  185. CSeq: 102 INVITE
  186. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  187. Date: Tue, 17 Jul 2012 16:19:35 GMT
  188. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  189. Supported: replaces, timer
  190. Content-Type: application/sdp
  191. Content-Length: 299
  192.  
  193. v=0
  194. o=root 47918836 47918836 IN IP4 192.168.80.1
  195. s=Asterisk PBX 1.8.11.0-1digium1~lucid
  196. c=IN IP4 192.168.80.1
  197. t=0 0
  198. m=audio 15220 RTP/AVP 8 0 101
  199. a=rtpmap:8 PCMA/8000
  200. a=rtpmap:0 PCMU/8000
  201. a=rtpmap:101 telephone-event/8000
  202. a=fmtp:101 0-16
  203. a=silenceSupp:off - - - -
  204. a=ptime:20
  205. a=sendrecv
  206.  
  207. ---
  208. [Jul 17 12:19:35] VERBOSE[24755] chan_sip.c:
  209. <--- Transmitting (NAT) to 208.73.146.95:5060 --->
  210. SIP/2.0 180 Ringing
  211. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1;received=208.73.146.95;rport=5060
  212. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  213. To: <sip:voip number@192.168.101.100:5060>;tag=as3f87b738
  214. Call-ID: pcst1342541974696254284310@192.168.201.113
  215. CSeq: 1 INVITE
  216. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  217. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  218. Supported: replaces, timer
  219. Session-Expires: 1800;refresher=uas
  220. Contact: <sip:voip number@173.167.212.105:5060>
  221. Content-Length: 0
  222.  
  223.  
  224. <------------>
  225. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  226. <--- SIP read from UDP:192.168.80.1:5061 --->
  227. SIP/2.0 100 Trying
  228. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport=5060
  229. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  230. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>
  231. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  232. CSeq: 102 INVITE
  233. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  234. Content-Length: 0
  235.  
  236. <------------->
  237. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  238. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  239. <--- SIP read from UDP:192.168.80.1:5061 --->
  240. SIP/2.0 101 Dialog Establishement
  241. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport=5060
  242. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  243. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=571506724
  244. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  245. CSeq: 102 INVITE
  246. Contact: <sip:200@173.167.212.105:5061>
  247. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  248. Content-Length: 0
  249.  
  250. <------------->
  251. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (9 headers 0 lines) ---
  252. [Jul 17 12:19:35] VERBOSE[24760] chan_sip.c: Really destroying SIP dialog '0298792d7fb4cd8b1922277e377b41be@192.168.80.1:5060' Method: INVITE
  253. [Jul 17 12:19:35] WARNING[24760] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  254. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Scheduling destruction of SIP dialog '3388268a14ebb44d16c44f41425fc3d3@192.168.80.1:5060' in 32000 ms (Method: NOTIFY)
  255. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  256. NOTIFY sip:202@192.168.80.131:5060 SIP/2.0
  257. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4bca5ab7;rport
  258. Max-Forwards: 70
  259. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as5a38967e
  260. To: <sip:202@192.168.80.131:5060>
  261. Contact: <sip:asterisk@192.168.80.1:5060>
  262. Call-ID: 3388268a14ebb44d16c44f41425fc3d3@192.168.80.1:5060
  263. CSeq: 102 NOTIFY
  264. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  265. Event: message-summary
  266. Content-Type: application/simple-message-summary
  267. Content-Length: 92
  268.  
  269. Messages-Waiting: no
  270. Message-Account: sip:asterisk@192.168.80.1
  271. Voice-Message: 0/0 (0/0)
  272.  
  273. ---
  274. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Scheduling destruction of SIP dialog '3932dbce2aa74d8e33a9d7840f76364e@192.168.80.1:5060' in 32000 ms (Method: NOTIFY)
  275. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  276. NOTIFY sip:201@192.168.80.131:5060 SIP/2.0
  277. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5e0ab7f4;rport
  278. Max-Forwards: 70
  279. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as1e916658
  280. To: <sip:201@192.168.80.131:5060>
  281. Contact: <sip:asterisk@192.168.80.1:5060>
  282. Call-ID: 3932dbce2aa74d8e33a9d7840f76364e@192.168.80.1:5060
  283. CSeq: 102 NOTIFY
  284. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  285. Event: message-summary
  286. Content-Type: application/simple-message-summary
  287. Content-Length: 92
  288.  
  289. Messages-Waiting: no
  290. Message-Account: sip:asterisk@192.168.80.1
  291. Voice-Message: 0/0 (0/0)
  292.  
  293. ---
  294. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Audio is at 10666
  295. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  296. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  297. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  298. [Jul 17 12:19:35] VERBOSE[24762] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  299. INVITE sip:202@192.168.80.131:5060 SIP/2.0
  300. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe;rport
  301. Max-Forwards: 70
  302. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  303. To: <sip:202@192.168.80.131:5060>
  304. Contact: <sip:voip number@192.168.80.1:5060>
  305. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  306. CSeq: 102 INVITE
  307. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  308. Date: Tue, 17 Jul 2012 16:19:35 GMT
  309. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  310. Supported: replaces, timer
  311. Content-Type: application/sdp
  312. Content-Length: 303
  313.  
  314. v=0
  315. o=root 1261129094 1261129094 IN IP4 192.168.80.1
  316. s=Asterisk PBX 1.8.11.0-1digium1~lucid
  317. c=IN IP4 192.168.80.1
  318. t=0 0
  319. m=audio 10666 RTP/AVP 8 0 101
  320. a=rtpmap:8 PCMA/8000
  321. a=rtpmap:0 PCMU/8000
  322. a=rtpmap:101 telephone-event/8000
  323. a=fmtp:101 0-16
  324. a=silenceSupp:off - - - -
  325. a=ptime:20
  326. a=sendrecv
  327.  
  328. ---
  329. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Audio is at 13566
  330. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  331. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  332. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  333. [Jul 17 12:19:35] VERBOSE[24761] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  334. INVITE sip:201@192.168.80.131:5060 SIP/2.0
  335. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6;rport
  336. Max-Forwards: 70
  337. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  338. To: <sip:201@192.168.80.131:5060>
  339. Contact: <sip:voip number@192.168.80.1:5060>
  340. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  341. CSeq: 102 INVITE
  342. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  343. Date: Tue, 17 Jul 2012 16:19:35 GMT
  344. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  345. Supported: replaces, timer
  346. Content-Type: application/sdp
  347. Content-Length: 303
  348.  
  349. v=0
  350. o=root 2071523935 2071523935 IN IP4 192.168.80.1
  351. s=Asterisk PBX 1.8.11.0-1digium1~lucid
  352. c=IN IP4 192.168.80.1
  353. t=0 0
  354. m=audio 13566 RTP/AVP 8 0 101
  355. a=rtpmap:8 PCMA/8000
  356. a=rtpmap:0 PCMU/8000
  357. a=rtpmap:101 telephone-event/8000
  358. a=fmtp:101 0-16
  359. a=silenceSupp:off - - - -
  360. a=ptime:20
  361. a=sendrecv
  362.  
  363. ---
  364. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  365. <--- SIP read from UDP:192.168.80.131:5060 --->
  366. SIP/2.0 200 OK
  367. To: <sip:202@192.168.80.131:5060>;tag=2c75c27d129ad19i1
  368. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as5a38967e
  369. Call-ID: 3388268a14ebb44d16c44f41425fc3d3@192.168.80.1:5060
  370. CSeq: 102 NOTIFY
  371. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4bca5ab7
  372. Server: RTP300-3.1.22
  373. Content-Length: 0
  374.  
  375. <------------->
  376. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  377. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog '3388268a14ebb44d16c44f41425fc3d3@192.168.80.1:5060' Method: NOTIFY
  378. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  379. <--- SIP read from UDP:192.168.80.131:5060 --->
  380. SIP/2.0 200 OK
  381. To: <sip:201@192.168.80.131:5060>;tag=20d025976021f269i0
  382. From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as1e916658
  383. Call-ID: 3932dbce2aa74d8e33a9d7840f76364e@192.168.80.1:5060
  384. CSeq: 102 NOTIFY
  385. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5e0ab7f4
  386. Server: RTP300-3.1.22
  387. Content-Length: 0
  388.  
  389. <------------->
  390. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  391. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog '3932dbce2aa74d8e33a9d7840f76364e@192.168.80.1:5060' Method: NOTIFY
  392. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  393. <--- SIP read from UDP:192.168.80.131:5060 --->
  394. SIP/2.0 100 Trying
  395. To: <sip:202@192.168.80.131:5060>
  396. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  397. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  398. CSeq: 102 INVITE
  399. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe
  400. Server: RTP300-3.1.22
  401. Content-Length: 0
  402.  
  403. <------------->
  404. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  405. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  406. <--- SIP read from UDP:192.168.80.131:5060 --->
  407. SIP/2.0 100 Trying
  408. To: <sip:201@192.168.80.131:5060>
  409. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  410. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  411. CSeq: 102 INVITE
  412. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6
  413. Server: RTP300-3.1.22
  414. Content-Length: 0
  415.  
  416. <------------->
  417. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  418. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  419. <--- SIP read from UDP:192.168.80.131:5060 --->
  420. SIP/2.0 180 Ringing
  421. To: <sip:202@192.168.80.131:5060>;tag=6b5101b785acd86ei1
  422. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  423. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  424. CSeq: 102 INVITE
  425. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe
  426. Server: RTP300-3.1.22
  427. Content-Length: 0
  428.  
  429. <------------->
  430. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  431. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: list_route: no route
  432. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c:
  433. <--- SIP read from UDP:192.168.80.131:5060 --->
  434. SIP/2.0 180 Ringing
  435. To: <sip:201@192.168.80.131:5060>;tag=33eb26174791ff8fi0
  436. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  437. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  438. CSeq: 102 INVITE
  439. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6
  440. Server: RTP300-3.1.22
  441. Content-Length: 0
  442.  
  443. <------------->
  444. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  445. [Jul 17 12:19:35] VERBOSE[2534] chan_sip.c: list_route: no route
  446. [Jul 17 12:19:36] VERBOSE[2534] chan_sip.c:
  447. <--- SIP read from UDP:192.168.80.1:5061 --->
  448. SIP/2.0 180 Ringing
  449. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport=5060
  450. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  451. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=571506724
  452. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  453. CSeq: 102 INVITE
  454. Contact: <sip:200@173.167.212.105:5061>
  455. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  456. Content-Length: 0
  457.  
  458. <------------->
  459. [Jul 17 12:19:36] VERBOSE[2534] chan_sip.c: --- (9 headers 0 lines) ---
  460. [Jul 17 12:19:36] VERBOSE[2534] chan_sip.c: list_route: hop: <sip:200@173.167.212.105:5061>
  461. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  462. <--- SIP read from UDP:208.73.146.95:5060 --->
  463. CANCEL sip:voip number@173.167.212.105:5060 SIP/2.0
  464. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1
  465. CSeq: 1 CANCEL
  466. Call-ID: pcst1342541974696254284310@192.168.201.113
  467. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  468. To: <sip:voip number@192.168.101.100:5060>
  469. Max-Forwards: 70
  470. Content-Length: 0
  471.  
  472. <------------->
  473. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  474. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Sending to 208.73.146.95:5060 (NAT)
  475. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  476. <--- Reliably Transmitting (NAT) to 208.73.146.95:5060 --->
  477. SIP/2.0 487 Request Terminated
  478. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1;received=208.73.146.95;rport=5060
  479. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  480. To: <sip:voip number@192.168.101.100:5060>;tag=as3f87b738
  481. Call-ID: pcst1342541974696254284310@192.168.201.113
  482. CSeq: 1 INVITE
  483. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  484. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  485. Supported: replaces, timer
  486. Content-Length: 0
  487.  
  488.  
  489. <------------>
  490. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  491. <--- Transmitting (NAT) to 208.73.146.95:5060 --->
  492. SIP/2.0 200 OK
  493. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1;received=208.73.146.95;rport=5060
  494. From: "ALOHN,CHOWS" <sip:+12153681640@192.168.101.113:5060>;tag=3551530774-462988
  495. To: <sip:voip number@192.168.101.100:5060>;tag=as3f87b738
  496. Call-ID: pcst1342541974696254284310@192.168.201.113
  497. CSeq: 1 CANCEL
  498. Server: Asterisk PBX 1.8.11.0-1digium1~lucid
  499. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  500. Supported: replaces, timer
  501. Content-Length: 0
  502.  
  503.  
  504. <------------>
  505. [Jul 17 12:19:40] VERBOSE[24763] chan_sip.c: Scheduling destruction of SIP dialog '45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  506. [Jul 17 12:19:40] VERBOSE[24763] chan_sip.c: set_destination: Parsing <sip:200@192.168.80.1:5061;line=70cc019d1600592> for address/port to send to
  507. [Jul 17 12:19:40] VERBOSE[24763] chan_sip.c: set_destination: set destination to 192.168.80.1:5061
  508. [Jul 17 12:19:40] VERBOSE[24763] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.1:5061:
  509. CANCEL sip:200@192.168.80.1:5061;line=70cc019d1600592 SIP/2.0
  510. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport
  511. Max-Forwards: 70
  512. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  513. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>
  514. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  515. CSeq: 102 CANCEL
  516. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  517. Content-Length: 0
  518.  
  519.  
  520. ---
  521. [Jul 17 12:19:40] VERBOSE[24763] chan_sip.c: Scheduling destruction of SIP dialog '45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  522. [Jul 17 12:19:40] VERBOSE[24762] chan_sip.c: Scheduling destruction of SIP dialog '7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  523. [Jul 17 12:19:40] VERBOSE[24762] chan_sip.c: set_destination: Parsing <sip:202@192.168.80.131:5060> for address/port to send to
  524. [Jul 17 12:19:40] VERBOSE[24762] chan_sip.c: set_destination: set destination to 192.168.80.131:5060
  525. [Jul 17 12:19:40] VERBOSE[24762] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  526. CANCEL sip:202@192.168.80.131:5060 SIP/2.0
  527. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe;rport
  528. Max-Forwards: 70
  529. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  530. To: <sip:202@192.168.80.131:5060>
  531. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  532. CSeq: 102 CANCEL
  533. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  534. Content-Length: 0
  535.  
  536.  
  537. ---
  538. [Jul 17 12:19:40] VERBOSE[24762] chan_sip.c: Scheduling destruction of SIP dialog '7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  539. [Jul 17 12:19:40] VERBOSE[24761] chan_sip.c: Scheduling destruction of SIP dialog '4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  540. [Jul 17 12:19:40] VERBOSE[24761] chan_sip.c: set_destination: Parsing <sip:201@192.168.80.131:5060> for address/port to send to
  541. [Jul 17 12:19:40] VERBOSE[24761] chan_sip.c: set_destination: set destination to 192.168.80.131:5060
  542. [Jul 17 12:19:40] VERBOSE[24761] chan_sip.c: Reliably Transmitting (NAT) to 192.168.80.131:5060:
  543. CANCEL sip:201@192.168.80.131:5060 SIP/2.0
  544. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6;rport
  545. Max-Forwards: 70
  546. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  547. To: <sip:201@192.168.80.131:5060>
  548. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  549. CSeq: 102 CANCEL
  550. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  551. Content-Length: 0
  552.  
  553.  
  554. ---
  555. [Jul 17 12:19:40] VERBOSE[24761] chan_sip.c: Scheduling destruction of SIP dialog '4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  556. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  557. <--- SIP read from UDP:192.168.80.1:5061 --->
  558. SIP/2.0 200 OK
  559. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport=5060
  560. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  561. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=571506724
  562. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  563. CSeq: 102 CANCEL
  564. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  565. Content-Length: 0
  566.  
  567. <------------->
  568. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  569. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  570. <--- SIP read from UDP:192.168.80.1:5061 --->
  571. SIP/2.0 487 Request Cancelled
  572. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport=5060
  573. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  574. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=571506724
  575. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  576. CSeq: 102 INVITE
  577. User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
  578. Content-Length: 0
  579.  
  580. <------------->
  581. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  582. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: Parsing <sip:200@192.168.80.1:5061;line=70cc019d1600592> for address/port to send to
  583. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: set destination to 192.168.80.1:5061
  584. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Transmitting (NAT) to 192.168.80.1:5061:
  585. ACK sip:200@173.167.212.105:5061 SIP/2.0
  586. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK7e07cc34;rport
  587. Max-Forwards: 70
  588. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4a93bd02
  589. To: <sip:200@192.168.80.1:5061;line=70cc019d1600592>;tag=571506724
  590. Contact: <sip:voip number@192.168.80.1:5060>
  591. Call-ID: 45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060
  592. CSeq: 102 ACK
  593. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  594. Content-Length: 0
  595.  
  596.  
  597. ---
  598. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Scheduling destruction of SIP dialog '45df8ac42ad11672212121b46d07d3e1@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  599. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  600. <--- SIP read from UDP:192.168.80.131:5060 --->
  601. SIP/2.0 487 Request Terminated
  602. To: <sip:201@192.168.80.131:5060>;tag=33eb26174791ff8fi0
  603. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  604. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  605. CSeq: 102 INVITE
  606. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6
  607. Server: RTP300-3.1.22
  608. Content-Length: 0
  609.  
  610. <------------->
  611. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  612. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: Parsing <sip:201@192.168.80.131:5060> for address/port to send to
  613. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: set destination to 192.168.80.131:5060
  614. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Transmitting (NAT) to 192.168.80.131:5060:
  615. ACK sip:201@192.168.80.131:5060 SIP/2.0
  616. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6;rport
  617. Max-Forwards: 70
  618. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  619. To: <sip:201@192.168.80.131:5060>;tag=33eb26174791ff8fi0
  620. Contact: <sip:voip number@192.168.80.1:5060>
  621. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  622. CSeq: 102 ACK
  623. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  624. Content-Length: 0
  625.  
  626.  
  627. ---
  628. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Scheduling destruction of SIP dialog '4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  629. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  630. <--- SIP read from UDP:192.168.80.131:5060 --->
  631. SIP/2.0 200 OK
  632. To: <sip:201@192.168.80.131:5060>;tag=33eb26174791ff8fi0
  633. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as04dc9efb
  634. Call-ID: 4a31977548f73177563c7c6e6b2979df@192.168.80.1:5060
  635. CSeq: 102 CANCEL
  636. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK74bc4bd6
  637. Server: RTP300-3.1.22
  638. Content-Length: 0
  639.  
  640. <------------->
  641. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  642. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  643. <--- SIP read from UDP:192.168.80.131:5060 --->
  644. SIP/2.0 487 Request Terminated
  645. To: <sip:202@192.168.80.131:5060>;tag=6b5101b785acd86ei1
  646. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  647. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  648. CSeq: 102 INVITE
  649. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe
  650. Server: RTP300-3.1.22
  651. Content-Length: 0
  652.  
  653. <------------->
  654. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  655. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: Parsing <sip:202@192.168.80.131:5060> for address/port to send to
  656. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: set_destination: set destination to 192.168.80.131:5060
  657. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Transmitting (NAT) to 192.168.80.131:5060:
  658. ACK sip:202@192.168.80.131:5060 SIP/2.0
  659. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe;rport
  660. Max-Forwards: 70
  661. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  662. To: <sip:202@192.168.80.131:5060>;tag=6b5101b785acd86ei1
  663. Contact: <sip:voip number@192.168.80.1:5060>
  664. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  665. CSeq: 102 ACK
  666. User-Agent: Asterisk PBX 1.8.11.0-1digium1~lucid
  667. Content-Length: 0
  668.  
  669.  
  670. ---
  671. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Scheduling destruction of SIP dialog '7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060' in 32000 ms (Method: INVITE)
  672. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  673. <--- SIP read from UDP:192.168.80.131:5060 --->
  674. SIP/2.0 200 OK
  675. To: <sip:202@192.168.80.131:5060>;tag=6b5101b785acd86ei1
  676. From: "ALOHN,CHOWS" <sip:voip number@192.168.80.1>;tag=as4921db60
  677. Call-ID: 7017bf7f388bcd7b4bad7ac2386657ef@192.168.80.1:5060
  678. CSeq: 102 CANCEL
  679. Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e85dffe
  680. Server: RTP300-3.1.22
  681. Content-Length: 0
  682.  
  683. <------------->
  684. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  685. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c:
  686. <--- SIP read from UDP:208.73.146.95:5060 --->
  687. ACK sip:2156605325@173.167.212.105:5060 SIP/2.0
  688. Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKtl6s560008mgds40h0a1.1
  689. CSeq: 1 ACK
  690. Call-ID: pcst1342541974696254284310@192.168.201.113
  691. From: "ALOHN,CHOWS" <sip+1 lane line number@192.168.101.113:5060>;tag=3551530774-462988
  692. To: <sip:voip number@192.168.101.100:5060>;tag=as3f87b738
  693. Max-Forwards: 70
  694. Content-Length: 0
  695.  
  696. <------------->
  697. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: --- (8 headers 0 lines) ---
  698. [Jul 17 12:19:40] VERBOSE[2534] chan_sip.c: Really destroying SIP dialog 'pcst1342541974696254284310@192.168.201.113' Method: ACK
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