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  1. <--- SIP read from UDP:87.20.161.118:50295 --->
  2.  
  3.  
  4. <------------->
  5.  
  6. <--- SIP read from UDP:10.0.8.2:56556 --->
  7.  
  8.  
  9. <------------->
  10.  
  11. <--- SIP read from UDP:10.0.8.2:56556 --->
  12. INVITE sip:333@thmdev.com SIP/2.0
  13. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjZGagYsrRAcRkOG.tDij810GLfJYOQ9gy
  14. Max-Forwards: 70
  15. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  16. To: <sip:333@thmdev.com>
  17. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  18. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  19. CSeq: 16012 INVITE
  20. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  21. upported: replaces, 100rel, timer, norefersub
  22. Session-Expires: 1800
  23. Min-SE: 90
  24. User-Agent: Telephone 1.0.2
  25. Content-Type: application/sdp
  26. Content-Length: 447
  27.  
  28. v=0
  29. o=- 3523968081 3523968081 IN IP4 10.0.8.2
  30. s=pjmedia
  31. c=IN IP4 10.0.8.2
  32. t=0 0
  33. a=X-nat:0
  34. m=audio 4010 RTP/AVP 103 102 104 109 3 0 8 9 101
  35. a=rtcp:4011 IN IP4 10.0.8.2
  36. a=rtpmap:103 speex/16000
  37. a=rtpmap:102 speex/8000
  38. a=rtpmap:104 speex/32000
  39. a=rtpmap:109 iLBC/8000
  40. a=fmtp:109 mode=30
  41. a=rtpmap:3 GSM/8000
  42. a=rtpmap:0 PCMU/8000
  43. a=rtpmap:8 PCMA/8000
  44. a=rtpmap:9 G722/8000
  45. a=sendrecv
  46. a=rtpmap:101 telephone-event/8000
  47. a=fmtp:101 0-15
  48.  
  49. <------------->
  50. --- (15 headers 20 lines) ---
  51. Sending to 10.0.8.2 : 56556 (no NAT)
  52. Using INVITE request as basis request - .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  53. Found peer 'simone' for 'simone' from 10.0.8.2:56556
  54.  
  55. <--- Reliably Transmitting (no NAT) to 10.0.8.2:56556 --->
  56. SIP/2.0 401 Unauthorized
  57. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjZGagYsrRAcRkOG.tDij810GLfJYOQ9gy;received=10.0.8.2;rport=56556
  58. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  59. To: <sip:333@thmdev.com>;tag=as22d610b4
  60. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  61. CSeq: 16012 INVITE
  62. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  64. Supported: replaces, timer
  65. WWW-Authenticate: Digest algorithm=MD5, realm="thmdev.com", nonce="0fe6f1ab"
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. Scheduling destruction of SIP dialog '.oxJQAngpz79SKuhNJQUGWOd0tYb6P0m' in 32000 ms (Method: INVITE)
  71.  
  72. <--- SIP read from UDP:10.0.8.2:56556 --->
  73. ACK sip:333@thmdev.com SIP/2.0
  74. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjZGagYsrRAcRkOG.tDij810GLfJYOQ9gy
  75. Max-Forwards: 70
  76. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  77. To: <sip:333@thmdev.com>;tag=as22d610b4
  78. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  79. CSeq: 16012 ACK
  80. Content-Length: 0
  81.  
  82.  
  83. <------------->
  84. --- (8 headers 0 lines) ---
  85.  
  86. <--- SIP read from UDP:10.0.8.2:56556 --->
  87. INVITE sip:333@thmdev.com SIP/2.0
  88. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjy1NypcRMH1aDcHWAnFIEjgBx4cvfHtKK
  89. Max-Forwards: 70
  90. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  91. To: <sip:333@thmdev.com>
  92. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  93. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  94. CSeq: 16013 INVITE
  95. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  96. upported: replaces, 100rel, timer, norefersub
  97. Session-Expires: 1800
  98. Min-SE: 90
  99. User-Agent: Telephone 1.0.2
  100. Authorization: Digest username="simone", realm="thmdev.com", nonce="0fe6f1ab", uri="sip:333@thmdev.com", response="f8f422b577571e468c1c03fcc5a69e50", algorithm=MD5
  101. Content-Type: application/sdp
  102. Content-Length: 447
  103.  
  104. v=0
  105. o=- 3523968081 3523968081 IN IP4 10.0.8.2
  106. s=pjmedia
  107. c=IN IP4 10.0.8.2
  108. t=0 0
  109. a=X-nat:0
  110. m=audio 4010 RTP/AVP 103 102 104 109 3 0 8 9 101
  111. a=rtcp:4011 IN IP4 10.0.8.2
  112. a=rtpmap:103 speex/16000
  113. a=rtpmap:102 speex/8000
  114. a=rtpmap:104 speex/32000
  115. a=rtpmap:109 iLBC/8000
  116. a=fmtp:109 mode=30
  117. a=rtpmap:3 GSM/8000
  118. a=rtpmap:0 PCMU/8000
  119. a=rtpmap:8 PCMA/8000
  120. a=rtpmap:9 G722/8000
  121. a=sendrecv
  122. a=rtpmap:101 telephone-event/8000
  123. a=fmtp:101 0-15
  124.  
  125. <------------->
  126. --- (16 headers 20 lines) ---
  127. Sending to 10.0.8.2 : 56556 (no NAT)
  128. Using INVITE request as basis request - .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  129. Found peer 'simone' for 'simone' from 10.0.8.2:56556
  130. Found RTP audio format 103
  131. Found RTP audio format 102
  132. Found RTP audio format 104
  133. Found RTP audio format 109
  134. Found RTP audio format 3
  135. Found RTP audio format 0
  136. Found RTP audio format 8
  137. Found RTP audio format 9
  138. Found RTP audio format 101
  139. Found audio description format speex for ID 103
  140. Found audio description format speex for ID 102
  141. Found audio description format speex for ID 104
  142. Found audio description format iLBC for ID 109
  143. Found audio description format GSM for ID 3
  144. Found audio description format PCMU for ID 0
  145. Found audio description format PCMA for ID 8
  146. Found audio description format G722 for ID 9
  147. Found audio description format telephone-event for ID 101
  148. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x50160e (gsm|ulaw|alaw|speex|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  149. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  150. Peer audio RTP is at port 10.0.8.2:4010
  151. Looking for 333 in metwit (domain thmdev.com)
  152. list_route: hop: <sip:simone@10.0.8.2:56556;ob>
  153.  
  154. <--- Transmitting (no NAT) to 10.0.8.2:56556 --->
  155. SIP/2.0 100 Trying
  156. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjy1NypcRMH1aDcHWAnFIEjgBx4cvfHtKK;received=10.0.8.2;rport=56556
  157. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  158. To: <sip:333@thmdev.com>
  159. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  160. CSeq: 16013 INVITE
  161. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  162. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  163. Supported: replaces, timer
  164. Require: timer
  165. Session-Expires: 1800;refresher=uas
  166. Contact: <sip:333@10.0.8.1>
  167. Content-Length: 0
  168.  
  169.  
  170. <------------>
  171. Audio is at 85.25.10.25 port 17724
  172. Adding codec 0x4 (ulaw) to SDP
  173. Adding codec 0x2 (gsm) to SDP
  174. Adding codec 0x8 (alaw) to SDP
  175. Adding non-codec 0x1 (telephone-event) to SDP
  176. Reliably Transmitting (NAT) to 87.20.161.118:50295:
  177. INVITE sip:davide@192.168.1.3:50295;ob SIP/2.0
  178. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK1d07fa71;rport
  179. Max-Forwards: 70
  180. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  181. To: <sip:davide@192.168.1.3:50295;ob>
  182. Contact: <sip:42@85.25.10.25>
  183. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  184. CSeq: 102 INVITE
  185. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  186. Date: Fri, 02 Sep 2011 16:01:19 GMT
  187. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  188. Supported: replaces, timer
  189. Content-Type: application/sdp
  190. Content-Length: 292
  191.  
  192. v=0
  193. o=root 1900233878 1900233878 IN IP4 85.25.10.25
  194. s=Asterisk PBX 1.6.2.9-2+squeeze3
  195. c=IN IP4 85.25.10.25
  196. t=0 0
  197. m=audio 17724 RTP/AVP 0 3 8 101
  198. a=rtpmap:0 PCMU/8000
  199. a=rtpmap:3 GSM/8000
  200. a=rtpmap:8 PCMA/8000
  201. a=rtpmap:101 telephone-event/8000
  202. a=fmtp:101 0-16
  203. a=ptime:20
  204. a=sendrecv
  205.  
  206. ---
  207.  
  208. <--- SIP read from UDP:87.20.161.118:50295 --->
  209. SIP/2.0 100 Trying
  210. Via: SIP/2.0/UDP 85.25.10.25:5060;rport=5060;received=85.25.10.25;branch=z9hG4bK1d07fa71
  211. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  212. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  213. To: <sip:davide@192.168.1.3;ob>
  214. CSeq: 102 INVITE
  215. Content-Length: 0
  216.  
  217.  
  218. <------------->
  219. --- (7 headers 0 lines) ---
  220.  
  221. <--- SIP read from UDP:87.20.161.118:50295 --->
  222. SIP/2.0 180 Ringing
  223. Via: SIP/2.0/UDP 85.25.10.25:5060;rport=5060;received=85.25.10.25;branch=z9hG4bK1d07fa71
  224. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  225. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  226. To: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  227. CSeq: 102 INVITE
  228. Contact: "Davide Rizzo" <sip:davide@192.168.1.3:50295;ob>
  229. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  230. Content-Length: 0
  231.  
  232.  
  233. <------------->
  234. --- (9 headers 0 lines) ---
  235.  
  236. <--- Transmitting (no NAT) to 10.0.8.2:56556 --->
  237. SIP/2.0 180 Ringing
  238. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjy1NypcRMH1aDcHWAnFIEjgBx4cvfHtKK;received=10.0.8.2;rport=56556
  239. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  240. To: <sip:333@thmdev.com>;tag=as1d0562b5
  241. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  242. CSeq: 16013 INVITE
  243. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  244. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  245. Supported: replaces, timer
  246. Require: timer
  247. Session-Expires: 1800;refresher=uas
  248. Contact: <sip:333@10.0.8.1>
  249. Content-Length: 0
  250.  
  251.  
  252. <------------>
  253.  
  254. <--- SIP read from UDP:87.20.161.118:50295 --->
  255. SIP/2.0 200 OK
  256. Via: SIP/2.0/UDP 85.25.10.25:5060;rport=5060;received=85.25.10.25;branch=z9hG4bK1d07fa71
  257. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  258. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  259. To: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  260. CSeq: 102 INVITE
  261. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  262. Contact: "Davide Rizzo" <sip:davide@192.168.1.3:50295;ob>
  263. upported: replaces, 100rel, timer, norefersub
  264. Content-Type: application/sdp
  265. Content-Length: 248
  266.  
  267. v=0
  268. o=- 3523968081 3523968082 IN IP4 192.168.1.3
  269. s=pjmedia
  270. c=IN IP4 192.168.1.3
  271. t=0 0
  272. a=X-nat:0
  273. m=audio 4006 RTP/AVP 0 101
  274. a=rtcp:4007 IN IP4 192.168.1.3
  275. a=rtpmap:0 PCMU/8000
  276. a=sendrecv
  277. a=rtpmap:101 telephone-event/8000
  278. a=fmtp:101 0-15
  279.  
  280. <------------->
  281. --- (11 headers 12 lines) ---
  282. Found RTP audio format 0
  283. Found RTP audio format 101
  284. Found audio description format PCMU for ID 0
  285. Found audio description format telephone-event for ID 101
  286. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  287. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  288. Peer audio RTP is at port 192.168.1.3:4006
  289. list_route: hop: <sip:davide@192.168.1.3:50295;ob>
  290. set_destination: Parsing <sip:davide@192.168.1.3:50295;ob> for address/port to send to
  291. set_destination: set destination to 192.168.1.3, port 50295
  292. Transmitting (NAT) to 87.20.161.118:50295:
  293. ACK sip:davide@192.168.1.3:50295;ob SIP/2.0
  294. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK6ebefb92;rport
  295. Max-Forwards: 70
  296. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  297. To: <sip:davide@192.168.1.3:50295;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  298. Contact: <sip:42@85.25.10.25>
  299. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  300. CSeq: 102 ACK
  301. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  302. Content-Length: 0
  303.  
  304.  
  305. ---
  306. Audio is at 10.0.8.1 port 19178
  307. Adding codec 0x2 (gsm) to SDP
  308. Adding codec 0x4 (ulaw) to SDP
  309. Adding codec 0x8 (alaw) to SDP
  310. Adding non-codec 0x1 (telephone-event) to SDP
  311.  
  312. <--- Reliably Transmitting (no NAT) to 10.0.8.2:56556 --->
  313. SIP/2.0 200 OK
  314. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjy1NypcRMH1aDcHWAnFIEjgBx4cvfHtKK;received=10.0.8.2;rport=56556
  315. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  316. To: <sip:333@thmdev.com>;tag=as1d0562b5
  317. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  318. CSeq: 16013 INVITE
  319. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  320. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  321. Supported: replaces, timer
  322. Require: timer
  323. Session-Expires: 1800;refresher=uas
  324. Contact: <sip:333@10.0.8.1>
  325. Content-Type: application/sdp
  326. Content-Length: 284
  327.  
  328. v=0
  329. o=root 527489825 527489825 IN IP4 10.0.8.1
  330. s=Asterisk PBX 1.6.2.9-2+squeeze3
  331. c=IN IP4 10.0.8.1
  332. t=0 0
  333. m=audio 19178 RTP/AVP 3 0 8 101
  334. a=rtpmap:3 GSM/8000
  335. a=rtpmap:0 PCMU/8000
  336. a=rtpmap:8 PCMA/8000
  337. a=rtpmap:101 telephone-event/8000
  338. a=fmtp:101 0-16
  339. a=ptime:20
  340. a=sendrecv
  341.  
  342. <------------>
  343. set_destination: Parsing <sip:davide@192.168.1.3:50295;ob> for address/port to send to
  344. set_destination: set destination to 192.168.1.3, port 50295
  345. Audio is at 85.25.10.25 port 17724
  346. Adding codec 0x4 (ulaw) to SDP
  347. Adding non-codec 0x1 (telephone-event) to SDP
  348. Reliably Transmitting (NAT) to 87.20.161.118:50295:
  349. INVITE sip:davide@192.168.1.3:50295;ob SIP/2.0
  350. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK53b6d858;rport
  351. Max-Forwards: 70
  352. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  353. To: <sip:davide@192.168.1.3:50295;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  354. Contact: <sip:42@85.25.10.25>
  355. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  356. CSeq: 103 INVITE
  357. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  358. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  359. Supported: replaces, timer
  360. X-asterisk-Info: SIP re-invite (External RTP bridge)
  361. Content-Type: application/sdp
  362. Content-Length: 238
  363.  
  364. v=0
  365. o=root 1900233878 1900233879 IN IP4 10.0.8.2
  366. s=Asterisk PBX 1.6.2.9-2+squeeze3
  367. c=IN IP4 10.0.8.2
  368. t=0 0
  369. m=audio 4010 RTP/AVP 0 101
  370. a=rtpmap:0 PCMU/8000
  371. a=rtpmap:101 telephone-event/8000
  372. a=fmtp:101 0-16
  373. a=ptime:20
  374. a=sendrecv
  375.  
  376. ---
  377.  
  378. <--- SIP read from UDP:87.20.161.118:50295 --->
  379. SIP/2.0 200 OK
  380. Via: SIP/2.0/UDP 85.25.10.25:5060;rport=5060;received=85.25.10.25;branch=z9hG4bK53b6d858
  381. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  382. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  383. To: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  384. CSeq: 103 INVITE
  385. Contact: "Davide Rizzo" <sip:davide@192.168.1.3:50295;ob>
  386. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  387. upported: replaces, 100rel, timer, norefersub
  388. Content-Type: application/sdp
  389. Content-Length: 248
  390.  
  391. v=0
  392. o=- 3523968081 3523968083 IN IP4 192.168.1.3
  393. s=pjmedia
  394. c=IN IP4 192.168.1.3
  395. t=0 0
  396. a=X-nat:0
  397. m=audio 4006 RTP/AVP 0 101
  398. a=rtcp:4007 IN IP4 192.168.1.3
  399. a=rtpmap:0 PCMU/8000
  400. a=sendrecv
  401. a=rtpmap:101 telephone-event/8000
  402. a=fmtp:101 0-15
  403.  
  404. <------------->
  405. --- (11 headers 12 lines) ---
  406. Found RTP audio format 0
  407. Found RTP audio format 101
  408. Found audio description format PCMU for ID 0
  409. Found audio description format telephone-event for ID 101
  410. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  411. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  412. Peer audio RTP is at port 192.168.1.3:4006
  413. set_destination: Parsing <sip:davide@192.168.1.3:50295;ob> for address/port to send to
  414. set_destination: set destination to 192.168.1.3, port 50295
  415. Transmitting (NAT) to 87.20.161.118:50295:
  416. ACK sip:davide@192.168.1.3:50295;ob SIP/2.0
  417. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK1f91a233;rport
  418. Max-Forwards: 70
  419. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  420. To: <sip:davide@192.168.1.3:50295;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  421. Contact: <sip:42@85.25.10.25>
  422. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  423. CSeq: 103 ACK
  424. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  425. Content-Length: 0
  426.  
  427.  
  428. ---
  429.  
  430. <--- SIP read from UDP:10.0.8.2:56556 --->
  431. ACK sip:333@10.0.8.1 SIP/2.0
  432. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjhCG.ncDrb.TLYZVu.3AXZcltC0trf3uk
  433. Max-Forwards: 70
  434. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  435. To: <sip:333@thmdev.com>;tag=as1d0562b5
  436. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  437. CSeq: 16013 ACK
  438. Content-Length: 0
  439.  
  440.  
  441. <------------->
  442. --- (8 headers 0 lines) ---
  443. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  444. set_destination: set destination to 10.0.8.2, port 56556
  445. Audio is at 10.0.8.1 port 19178
  446. Adding codec 0x2 (gsm) to SDP
  447. Adding codec 0x4 (ulaw) to SDP
  448. Adding codec 0x8 (alaw) to SDP
  449. Adding non-codec 0x1 (telephone-event) to SDP
  450. Reliably Transmitting (no NAT) to 10.0.8.2:56556:
  451. INVITE sip:simone@10.0.8.2:56556;ob SIP/2.0
  452. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK18014656;rport
  453. Max-Forwards: 70
  454. From: <sip:333@thmdev.com>;tag=as1d0562b5
  455. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  456. Contact: <sip:333@10.0.8.1>
  457. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  458. CSeq: 102 INVITE
  459. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  460. Require: timer
  461. Session-Expires: 1800;refresher=uas
  462. Min-SE: 90
  463. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  464. Supported: replaces, timer
  465. X-asterisk-Info: SIP re-invite (External RTP bridge)
  466. Content-Type: application/sdp
  467. Content-Length: 289
  468.  
  469. v=0
  470. o=root 527489825 527489826 IN IP4 192.168.1.3
  471. s=Asterisk PBX 1.6.2.9-2+squeeze3
  472. c=IN IP4 192.168.1.3
  473. t=0 0
  474. m=audio 4006 RTP/AVP 3 0 8 101
  475. a=rtpmap:3 GSM/8000
  476. a=rtpmap:0 PCMU/8000
  477. a=rtpmap:8 PCMA/8000
  478. a=rtpmap:101 telephone-event/8000
  479. a=fmtp:101 0-16
  480. a=ptime:20
  481. a=sendrecv
  482.  
  483. ---
  484.  
  485. <--- SIP read from UDP:10.0.8.2:56556 --->
  486. INVITE sip:333@10.0.8.1 SIP/2.0
  487. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjPNpChm0fl-3UaqvHF1NRs1fpuIyw3g94
  488. Max-Forwards: 70
  489. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  490. To: <sip:333@thmdev.com>;tag=as1d0562b5
  491. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  492. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  493. CSeq: 16014 INVITE
  494. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  495. Supported: replaces, 100rel, timer, norefersub
  496. Session-Expires: 1800;refresher=uas
  497. Min-SE: 90
  498. Content-Type: application/sdp
  499. Content-Length: 238
  500.  
  501. v=0
  502. o=- 3523968081 3523968082 IN IP4 10.0.8.2
  503. s=pjmedia
  504. c=IN IP4 10.0.8.2
  505. t=0 0
  506. a=X-nat:0
  507. m=audio 4010 RTP/AVP 3 101
  508. a=rtcp:4011 IN IP4 10.0.8.2
  509. a=rtpmap:3 GSM/8000
  510. a=sendrecv
  511. a=rtpmap:101 telephone-event/8000
  512. a=fmtp:101 0-15
  513.  
  514. <------------->
  515. --- (14 headers 12 lines) ---
  516.  
  517. <--- Reliably Transmitting (no NAT) to 10.0.8.2:56556 --->
  518. SIP/2.0 491 Request Pending
  519. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjPNpChm0fl-3UaqvHF1NRs1fpuIyw3g94;received=10.0.8.2;rport=56556
  520. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  521. To: <sip:333@thmdev.com>;tag=as1d0562b5
  522. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  523. CSeq: 16014 INVITE
  524. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  525. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  526. Supported: replaces, timer
  527. Require: timer
  528. Session-Expires: 1800;refresher=uas
  529. Content-Length: 0
  530. X-Asterisk-HangupCause: Normal Clearing
  531. X-Asterisk-HangupCauseCode: 16
  532.  
  533.  
  534. <------------>
  535.  
  536. <--- SIP read from UDP:10.0.8.2:56556 --->
  537. SIP/2.0 491 Another INVITE transaction in progress
  538. Via: SIP/2.0/UDP 10.0.8.1:5060;rport=5060;received=10.0.8.1;branch=z9hG4bK18014656
  539. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  540. From: <sip:333@thmdev.com>;tag=as1d0562b5
  541. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  542. CSeq: 102 INVITE
  543. Content-Length: 0
  544.  
  545.  
  546. <------------->
  547. --- (7 headers 0 lines) ---
  548. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  549. set_destination: set destination to 10.0.8.2, port 56556
  550. Transmitting (no NAT) to 10.0.8.2:56556:
  551. ACK sip:simone@10.0.8.2:56556;ob SIP/2.0
  552. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK18014656;rport
  553. Max-Forwards: 70
  554. From: <sip:333@thmdev.com>;tag=as1d0562b5
  555. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  556. Contact: <sip:333@10.0.8.1>
  557. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  558. CSeq: 102 ACK
  559. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  560. Content-Length: 0
  561.  
  562.  
  563. ---
  564. [Sep 2 18:01:25] WARNING[29862]: chan_sip.c:18082 handle_response_invite: just did sched_add waitid(215) for sip_reinvite_retry for dialog .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m in handle_response_invite
  565.  
  566. <--- SIP read from UDP:10.0.8.2:56556 --->
  567. ACK sip:333@10.0.8.1 SIP/2.0
  568. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjPNpChm0fl-3UaqvHF1NRs1fpuIyw3g94
  569. Max-Forwards: 70
  570. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  571. To: <sip:333@thmdev.com>;tag=as1d0562b5
  572. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  573. CSeq: 16014 ACK
  574. Content-Length: 0
  575.  
  576.  
  577. <------------->
  578. --- (8 headers 0 lines) ---
  579. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  580. set_destination: set destination to 10.0.8.2, port 56556
  581. Audio is at 10.0.8.1 port 19178
  582. Adding codec 0x2 (gsm) to SDP
  583. Adding codec 0x4 (ulaw) to SDP
  584. Adding codec 0x8 (alaw) to SDP
  585. Adding non-codec 0x1 (telephone-event) to SDP
  586. Reliably Transmitting (no NAT) to 10.0.8.2:56556:
  587. INVITE sip:simone@10.0.8.2:56556;ob SIP/2.0
  588. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK6f17eb36;rport
  589. Max-Forwards: 70
  590. From: <sip:333@thmdev.com>;tag=as1d0562b5
  591. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  592. Contact: <sip:333@10.0.8.1>
  593. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  594. CSeq: 103 INVITE
  595. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  596. Require: timer
  597. Session-Expires: 1800;refresher=uas
  598. Min-SE: 90
  599. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  600. Supported: replaces, timer
  601. X-asterisk-Info: SIP re-invite (External RTP bridge)
  602. Content-Type: application/sdp
  603. Content-Length: 289
  604.  
  605. v=0
  606. o=root 527489825 527489827 IN IP4 192.168.1.3
  607. s=Asterisk PBX 1.6.2.9-2+squeeze3
  608. c=IN IP4 192.168.1.3
  609. t=0 0
  610. m=audio 4006 RTP/AVP 3 0 8 101
  611. a=rtpmap:3 GSM/8000
  612. a=rtpmap:0 PCMU/8000
  613. a=rtpmap:8 PCMA/8000
  614. a=rtpmap:101 telephone-event/8000
  615. a=fmtp:101 0-16
  616. a=ptime:20
  617. a=sendrecv
  618.  
  619. ---
  620.  
  621. <--- SIP read from UDP:10.0.8.2:56556 --->
  622. SIP/2.0 200 OK
  623. Via: SIP/2.0/UDP 10.0.8.1:5060;rport=5060;received=10.0.8.1;branch=z9hG4bK6f17eb36
  624. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  625. From: <sip:333@thmdev.com>;tag=as1d0562b5
  626. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  627. CSeq: 103 INVITE
  628. Session-Expires: 1800;refresher=uas
  629. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  630. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  631. Supported: replaces, 100rel, timer, norefersub
  632. Content-Type: application/sdp
  633. Content-Length: 238
  634.  
  635. v=0
  636. o=- 3523968081 3523968082 IN IP4 10.0.8.2
  637. s=pjmedia
  638. c=IN IP4 10.0.8.2
  639. t=0 0
  640. a=X-nat:0
  641. m=audio 4010 RTP/AVP 3 101
  642. a=rtcp:4011 IN IP4 10.0.8.2
  643. a=rtpmap:3 GSM/8000
  644. a=sendrecv
  645. a=rtpmap:101 telephone-event/8000
  646. a=fmtp:101 0-15
  647.  
  648. <------------->
  649. --- (12 headers 12 lines) ---
  650. Found RTP audio format 3
  651. Found RTP audio format 101
  652. Found audio description format GSM for ID 3
  653. Found audio description format telephone-event for ID 101
  654. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  655. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  656. Peer audio RTP is at port 10.0.8.2:4010
  657. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  658. set_destination: set destination to 10.0.8.2, port 56556
  659. Transmitting (no NAT) to 10.0.8.2:56556:
  660. ACK sip:simone@10.0.8.2:56556;ob SIP/2.0
  661. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK30c5fa5b;rport
  662. Max-Forwards: 70
  663. From: <sip:333@thmdev.com>;tag=as1d0562b5
  664. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  665. Contact: <sip:333@10.0.8.1>
  666. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  667. CSeq: 103 ACK
  668. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  669. Content-Length: 0
  670.  
  671.  
  672. ---
  673. set_destination: Parsing <sip:davide@192.168.1.3:50295;ob> for address/port to send to
  674. set_destination: set destination to 192.168.1.3, port 50295
  675. Audio is at 85.25.10.25 port 17724
  676. Adding codec 0x4 (ulaw) to SDP
  677. Adding non-codec 0x1 (telephone-event) to SDP
  678. Reliably Transmitting (NAT) to 87.20.161.118:50295:
  679. INVITE sip:davide@192.168.1.3:50295;ob SIP/2.0
  680. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK36e4247f;rport
  681. Max-Forwards: 70
  682. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  683. To: <sip:davide@192.168.1.3:50295;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  684. Contact: <sip:42@85.25.10.25>
  685. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  686. CSeq: 104 INVITE
  687. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  688. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  689. Supported: replaces, timer
  690. X-asterisk-Info: SIP re-invite (External RTP bridge)
  691. Content-Type: application/sdp
  692. Content-Length: 238
  693.  
  694. v=0
  695. o=root 1900233878 1900233880 IN IP4 10.0.8.2
  696. s=Asterisk PBX 1.6.2.9-2+squeeze3
  697. c=IN IP4 10.0.8.2
  698. t=0 0
  699. m=audio 4010 RTP/AVP 0 101
  700. a=rtpmap:0 PCMU/8000
  701. a=rtpmap:101 telephone-event/8000
  702. a=fmtp:101 0-16
  703. a=ptime:20
  704. a=sendrecv
  705.  
  706. ---
  707.  
  708. <--- SIP read from UDP:87.20.161.118:50295 --->
  709.  
  710.  
  711. <------------->
  712.  
  713. <--- SIP read from UDP:87.20.161.118:50295 --->
  714. SIP/2.0 200 OK
  715. Via: SIP/2.0/UDP 85.25.10.25:5060;rport=5060;received=85.25.10.25;branch=z9hG4bK36e4247f
  716. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  717. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  718. To: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  719. CSeq: 104 INVITE
  720. Contact: "Davide Rizzo" <sip:davide@192.168.1.3:50295;ob>
  721. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  722. upported: replaces, 100rel, timer, norefersub
  723. Content-Type: application/sdp
  724. Content-Length: 248
  725.  
  726. v=0
  727. o=- 3523968081 3523968084 IN IP4 192.168.1.3
  728. s=pjmedia
  729. c=IN IP4 192.168.1.3
  730. t=0 0
  731. a=X-nat:0
  732. m=audio 4006 RTP/AVP 0 101
  733. a=rtcp:4007 IN IP4 192.168.1.3
  734. a=rtpmap:0 PCMU/8000
  735. a=sendrecv
  736. a=rtpmap:101 telephone-event/8000
  737. a=fmtp:101 0-15
  738.  
  739. <------------->
  740. --- (11 headers 12 lines) ---
  741. Found RTP audio format 0
  742. Found RTP audio format 101
  743. Found audio description format PCMU for ID 0
  744. Found audio description format telephone-event for ID 101
  745. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  746. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  747. Peer audio RTP is at port 192.168.1.3:4006
  748. set_destination: Parsing <sip:davide@192.168.1.3:50295;ob> for address/port to send to
  749. set_destination: set destination to 192.168.1.3, port 50295
  750. Transmitting (NAT) to 87.20.161.118:50295:
  751. ACK sip:davide@192.168.1.3:50295;ob SIP/2.0
  752. Via: SIP/2.0/UDP 85.25.10.25:5060;branch=z9hG4bK46f48531;rport
  753. Max-Forwards: 70
  754. From: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  755. To: <sip:davide@192.168.1.3:50295;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  756. Contact: <sip:42@85.25.10.25>
  757. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  758. CSeq: 104 ACK
  759. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  760. Content-Length: 0
  761.  
  762.  
  763. ---
  764.  
  765. <--- SIP read from UDP:10.0.8.2:56556 --->
  766.  
  767.  
  768. <------------->
  769.  
  770. <--- SIP read from UDP:87.20.161.118:50295 --->
  771.  
  772.  
  773. <------------->
  774.  
  775. <--- SIP read from UDP:10.0.8.2:56556 --->
  776.  
  777.  
  778. <------------->
  779.  
  780. <--- SIP read from UDP:87.20.161.118:50295 --->
  781.  
  782.  
  783. <------------->
  784.  
  785. <--- SIP read from UDP:10.0.8.2:56556 --->
  786.  
  787.  
  788. <------------->
  789.  
  790. <--- SIP read from UDP:87.20.161.118:50295 --->
  791. BYE sip:42@85.25.10.25 SIP/2.0
  792. Via: SIP/2.0/UDP 192.168.1.3:50295;rport;branch=z9hG4bKPjlfsKUTpc-tyclYMzsImAAM4JnGrsBfBk
  793. Max-Forwards: 70
  794. From: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  795. To: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  796. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  797. CSeq: 1207 BYE
  798. User-Agent: Telephone 1.0.2
  799. Content-Length: 0
  800.  
  801.  
  802. <------------->
  803. --- (9 headers 0 lines) ---
  804. Sending to 87.20.161.118 : 50295 (NAT)
  805.  
  806. <--- Transmitting (NAT) to 87.20.161.118:50295 --->
  807. SIP/2.0 200 OK
  808. Via: SIP/2.0/UDP 192.168.1.3:50295;branch=z9hG4bKPjlfsKUTpc-tyclYMzsImAAM4JnGrsBfBk;received=87.20.161.118;rport=50295
  809. From: <sip:davide@192.168.1.3;ob>;tag=HHoPbWUDju1pjQDobxKFPMU5ZXc19Klw
  810. To: "Simone D'Amico" <sip:42@85.25.10.25>;tag=as7285932f
  811. Call-ID: 2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25
  812. CSeq: 1207 BYE
  813. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  814. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  815. Supported: replaces, timer
  816. Content-Length: 0
  817.  
  818.  
  819. <------------>
  820. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  821. set_destination: set destination to 10.0.8.2, port 56556
  822. Audio is at 10.0.8.1 port 19178
  823. Adding codec 0x2 (gsm) to SDP
  824. Adding non-codec 0x1 (telephone-event) to SDP
  825. Reliably Transmitting (no NAT) to 10.0.8.2:56556:
  826. INVITE sip:simone@10.0.8.2:56556;ob SIP/2.0
  827. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK38a55d54;rport
  828. Max-Forwards: 70
  829. From: <sip:333@thmdev.com>;tag=as1d0562b5
  830. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  831. Contact: <sip:333@10.0.8.1>
  832. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  833. CSeq: 104 INVITE
  834. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  835. Require: timer
  836. Session-Expires: 1800;refresher=uas
  837. Min-SE: 90
  838. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  839. Supported: replaces, timer
  840. X-asterisk-Info: SIP re-invite (External RTP bridge)
  841. Content-Type: application/sdp
  842. Content-Length: 236
  843.  
  844. v=0
  845. o=root 527489825 527489828 IN IP4 10.0.8.1
  846. s=Asterisk PBX 1.6.2.9-2+squeeze3
  847. c=IN IP4 10.0.8.1
  848. t=0 0
  849. m=audio 19178 RTP/AVP 3 101
  850. a=rtpmap:3 GSM/8000
  851. a=rtpmap:101 telephone-event/8000
  852. a=fmtp:101 0-16
  853. a=ptime:20
  854. a=sendrecv
  855.  
  856. ---
  857. Scheduling destruction of SIP dialog '.oxJQAngpz79SKuhNJQUGWOd0tYb6P0m' in 32000 ms (Method: ACK)
  858.  
  859. <--- SIP read from UDP:10.0.8.2:56556 --->
  860. SIP/2.0 200 OK
  861. Via: SIP/2.0/UDP 10.0.8.1:5060;rport=5060;received=10.0.8.1;branch=z9hG4bK38a55d54
  862. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  863. From: <sip:333@thmdev.com>;tag=as1d0562b5
  864. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  865. CSeq: 104 INVITE
  866. Session-Expires: 1800;refresher=uas
  867. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  868. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  869. Supported: replaces, 100rel, timer, norefersub
  870. Content-Type: application/sdp
  871. Content-Length: 238
  872.  
  873. v=0
  874. o=- 3523968081 3523968083 IN IP4 10.0.8.2
  875. s=pjmedia
  876. c=IN IP4 10.0.8.2
  877. t=0 0
  878. a=X-nat:0
  879. m=audio 4010 RTP/AVP 3 101
  880. a=rtcp:4011 IN IP4 10.0.8.2
  881. a=rtpmap:3 GSM/8000
  882. a=sendrecv
  883. a=rtpmap:101 telephone-event/8000
  884. a=fmtp:101 0-15
  885.  
  886. <------------->
  887. --- (12 headers 12 lines) ---
  888. Found RTP audio format 3
  889. Found RTP audio format 101
  890. Found audio description format GSM for ID 3
  891. Found audio description format telephone-event for ID 101
  892. Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  893. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  894. Peer audio RTP is at port 10.0.8.2:4010
  895. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  896. set_destination: set destination to 10.0.8.2, port 56556
  897. Transmitting (no NAT) to 10.0.8.2:56556:
  898. ACK sip:simone@10.0.8.2:56556;ob SIP/2.0
  899. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK051ee984;rport
  900. Max-Forwards: 70
  901. From: <sip:333@thmdev.com>;tag=as1d0562b5
  902. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  903. Contact: <sip:333@10.0.8.1>
  904. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  905. CSeq: 104 ACK
  906. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  907. Content-Length: 0
  908.  
  909.  
  910. ---
  911. set_destination: Parsing <sip:simone@10.0.8.2:56556;ob> for address/port to send to
  912. set_destination: set destination to 10.0.8.2, port 56556
  913. Reliably Transmitting (no NAT) to 10.0.8.2:56556:
  914. BYE sip:simone@10.0.8.2:56556;ob SIP/2.0
  915. Via: SIP/2.0/UDP 10.0.8.1:5060;branch=z9hG4bK26ddb39a;rport
  916. Max-Forwards: 70
  917. From: <sip:333@thmdev.com>;tag=as1d0562b5
  918. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  919. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  920. CSeq: 105 BYE
  921. User-Agent: Asterisk PBX 1.6.2.9-2+squeeze3
  922. X-Asterisk-HangupCause: Normal Clearing
  923. X-Asterisk-HangupCauseCode: 16
  924. Content-Length: 0
  925.  
  926.  
  927. ---
  928. Scheduling destruction of SIP dialog '.oxJQAngpz79SKuhNJQUGWOd0tYb6P0m' in 32000 ms (Method: ACK)
  929. Really destroying SIP dialog '2755e3bc1edc94f245a6d61232ce1b4d@85.25.10.25' Method: BYE
  930.  
  931. <--- SIP read from UDP:10.0.8.2:56556 --->
  932. SIP/2.0 200 OK
  933. Via: SIP/2.0/UDP 10.0.8.1:5060;rport=5060;received=10.0.8.1;branch=z9hG4bK26ddb39a
  934. Call-ID: .oxJQAngpz79SKuhNJQUGWOd0tYb6P0m
  935. From: <sip:333@thmdev.com>;tag=as1d0562b5
  936. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=BnS8MwYrypyTPS2T46o3cZCHtCTLP7Lz
  937. CSeq: 105 BYE
  938. Content-Length: 0
  939.  
  940.  
  941. <------------->
  942. --- (7 headers 0 lines) ---
  943. Really destroying SIP dialog '.oxJQAngpz79SKuhNJQUGWOd0tYb6P0m' Method: ACK
  944.  
  945. <--- SIP read from UDP:87.20.161.118:50295 --->
  946.  
  947.  
  948. <------------->
  949.  
  950. <--- SIP read from UDP:10.0.8.2:56556 --->
  951.  
  952.  
  953. <------------->
  954.  
  955. <--- SIP read from UDP:87.20.161.118:50295 --->
  956.  
  957.  
  958. <------------->
  959.  
  960. <--- SIP read from UDP:10.0.8.2:56556 --->
  961.  
  962.  
  963. <------------->
  964.  
  965. <--- SIP read from UDP:10.0.8.2:56556 --->
  966. REGISTER sip:thmdev.com SIP/2.0
  967. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjJVsUKYCXUhgV.1uka.R.CJotTy2vN..u
  968. Max-Forwards: 70
  969. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=jZkrytOHZXZBFGZCx9Y-fSndEebpNz7R
  970. To: "Simone D'Amico" <sip:simone@thmdev.com>
  971. Call-ID: 7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-
  972. CSeq: 38982 REGISTER
  973. User-Agent: Telephone 1.0.2
  974. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  975. Expires: 300
  976. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  977. Content-Length: 0
  978.  
  979.  
  980. <------------->
  981. --- (12 headers 0 lines) ---
  982. Sending to 10.0.8.2 : 56556 (no NAT)
  983.  
  984. <--- Transmitting (no NAT) to 10.0.8.2:56556 --->
  985. SIP/2.0 401 Unauthorized
  986. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjJVsUKYCXUhgV.1uka.R.CJotTy2vN..u;received=10.0.8.2;rport=56556
  987. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=jZkrytOHZXZBFGZCx9Y-fSndEebpNz7R
  988. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=as073a2a85
  989. Call-ID: 7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-
  990. CSeq: 38982 REGISTER
  991. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  992. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  993. Supported: replaces, timer
  994. WWW-Authenticate: Digest algorithm=MD5, realm="thmdev.com", nonce="36f0d68d"
  995. Content-Length: 0
  996.  
  997.  
  998. <------------>
  999. Scheduling destruction of SIP dialog '7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-' in 32000 ms (Method: REGISTER)
  1000.  
  1001. <--- SIP read from UDP:10.0.8.2:56556 --->
  1002. REGISTER sip:thmdev.com SIP/2.0
  1003. Via: SIP/2.0/UDP 10.0.8.2:56556;rport;branch=z9hG4bKPjZdaPc8k2Hytj6cl6cSxmdDrLv.25.5Wa
  1004. Max-Forwards: 70
  1005. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=jZkrytOHZXZBFGZCx9Y-fSndEebpNz7R
  1006. To: "Simone D'Amico" <sip:simone@thmdev.com>
  1007. Call-ID: 7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-
  1008. CSeq: 38983 REGISTER
  1009. User-Agent: Telephone 1.0.2
  1010. Contact: "Simone D'Amico" <sip:simone@10.0.8.2:56556;ob>
  1011. Expires: 300
  1012. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  1013. Authorization: Digest username="simone", realm="thmdev.com", nonce="36f0d68d", uri="sip:thmdev.com", response="a9924a293c5ee2bd59c6fd2d9cbe5dfb", algorithm=MD5
  1014. Content-Length: 0
  1015.  
  1016.  
  1017. <------------->
  1018. --- (13 headers 0 lines) ---
  1019. Sending to 10.0.8.2 : 56556 (no NAT)
  1020.  
  1021. <--- Transmitting (no NAT) to 10.0.8.2:56556 --->
  1022. SIP/2.0 200 OK
  1023. Via: SIP/2.0/UDP 10.0.8.2:56556;branch=z9hG4bKPjZdaPc8k2Hytj6cl6cSxmdDrLv.25.5Wa;received=10.0.8.2;rport=56556
  1024. From: "Simone D'Amico" <sip:simone@thmdev.com>;tag=jZkrytOHZXZBFGZCx9Y-fSndEebpNz7R
  1025. To: "Simone D'Amico" <sip:simone@thmdev.com>;tag=as073a2a85
  1026. Call-ID: 7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-
  1027. CSeq: 38983 REGISTER
  1028. Server: Asterisk PBX 1.6.2.9-2+squeeze3
  1029. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1030. Supported: replaces, timer
  1031. Expires: 300
  1032. Contact: <sip:simone@10.0.8.2:56556;ob>;expires=300
  1033. Date: Fri, 02 Sep 2011 16:02:39 GMT
  1034. Content-Length: 0
  1035.  
  1036.  
  1037. <------------>
  1038. Scheduling destruction of SIP dialog '7TTbNEReqwDpffcbzUIn-5nqCpLfSHY-' in 32000 ms (Method: REGISTER)
  1039.  
  1040. <--- SIP read from UDP:87.20.161.118:50295 --->
  1041.  
  1042.  
  1043. <------------->
  1044.  
  1045. <--- SIP read from UDP:10.0.8.2:56556 --->
  1046.  
  1047.  
  1048. <------------->
  1049.  
  1050. <--- SIP read from UDP:87.20.161.118:50295 --->
  1051.  
  1052.  
  1053. <------------->
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