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  1.  
  2. Appel d'un sip interne vers un portable externe via trunk EASYVOIP,
  3.  
  4. sonnerie OK, conversation OK, poste SIP raccroche OK
  5.  
  6. <--- SIP read from UDP:192.168.1.150:5060 --->
  7. INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  8. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK797cba3f2512db22
  9. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  10. To: <sip:06XXXXXXXX@192.168.1.200>
  11. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  12. Supported: replaces, timer, path
  13. X-Grandstream-PBX: true
  14. P-Early-Media: Supported
  15. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  16. CSeq: 42846 INVITE
  17. User-Agent: Grandstream GXP1200 1.2.5.3
  18. Max-Forwards: 70
  19. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  20. Content-Type: application/sdp
  21. Content-Length: 313
  22.  
  23. v=0
  24. o=101 8000 8000 IN IP4 192.168.1.150
  25. s=SIP Call
  26. c=IN IP4 192.168.1.150
  27. t=0 0
  28. m=audio 5010 RTP/AVP 9 18 8 0 2 101
  29. a=sendrecv
  30. a=rtpmap:9 G722/8000
  31. a=rtpmap:18 G729/8000
  32. a=rtpmap:8 PCMA/8000
  33. a=rtpmap:0 PCMU/8000
  34. a=rtpmap:2 G726-32/8000
  35. a=ptime:20
  36. a=rtpmap:101 telephone-event/8000
  37. a=fmtp:101 0-11
  38. <------------->
  39. --- (15 headers 15 lines) ---
  40. Sending to 192.168.1.150:5060 (no NAT)
  41. Using INVITE request as basis request - 95a53eef3afeb0c2@192.168.1.150
  42. Found peer '101' for '101' from 192.168.1.150:5060
  43.  
  44. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  45. SIP/2.0 401 Unauthorized
  46. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK797cba3f2512db22;received=192.168.1.150;rport=5060
  47. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  48. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as15c2ed34
  49. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  50. CSeq: 42846 INVITE
  51. Server: AskoziaPBX
  52. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  53. Supported: replaces, timer
  54. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ab59689"
  55. Content-Length: 0
  56.  
  57.  
  58. <------------>
  59. Scheduling destruction of SIP dialog '95a53eef3afeb0c2@192.168.1.150' in 6400 ms (Method: INVITE)
  60.  
  61. <--- SIP read from UDP:192.168.1.150:5060 --->
  62. ACK sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  63. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK797cba3f2512db22
  64. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  65. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as15c2ed34
  66. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  67. Supported: path
  68. X-Grandstream-PBX: true
  69. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  70. CSeq: 42846 ACK
  71. User-Agent: Grandstream GXP1200 1.2.5.3
  72. Max-Forwards: 70
  73. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  74. Content-Length: 0
  75.  
  76. <------------->
  77. --- (13 headers 0 lines) ---
  78.  
  79. <--- SIP read from UDP:192.168.1.150:5060 --->
  80. INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  81. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7b5cb758ecd1abce
  82. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  83. To: <sip:06XXXXXXXX@192.168.1.200>
  84. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  85. Supported: replaces, timer, path
  86. X-Grandstream-PBX: true
  87. P-Early-Media: Supported
  88. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="1ab59689", response="e9789d0f23f0c051e85b399e2867409d"
  89. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  90. CSeq: 42847 INVITE
  91. User-Agent: Grandstream GXP1200 1.2.5.3
  92. Max-Forwards: 70
  93. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  94. Content-Type: application/sdp
  95. Content-Length: 313
  96.  
  97. v=0
  98. o=101 8000 8001 IN IP4 192.168.1.150
  99. s=SIP Call
  100. c=IN IP4 192.168.1.150
  101. t=0 0
  102. m=audio 5010 RTP/AVP 9 18 8 0 2 101
  103. a=sendrecv
  104. a=rtpmap:9 G722/8000
  105. a=rtpmap:18 G729/8000
  106. a=rtpmap:8 PCMA/8000
  107. a=rtpmap:0 PCMU/8000
  108. a=rtpmap:2 G726-32/8000
  109. a=ptime:20
  110. a=rtpmap:101 telephone-event/8000
  111. a=fmtp:101 0-11
  112. <------------->
  113. --- (16 headers 15 lines) ---
  114. Sending to 192.168.1.150:5060 (NAT)
  115. Using INVITE request as basis request - 95a53eef3afeb0c2@192.168.1.150
  116. Found peer '101' for '101' from 192.168.1.150:5060
  117. Found RTP audio format 9
  118. Found RTP audio format 18
  119. Found RTP audio format 8
  120. Found RTP audio format 0
  121. Found RTP audio format 2
  122. Found RTP audio format 101
  123. Found audio description format G722 for ID 9
  124. Found audio description format G729 for ID 18
  125. Found audio description format PCMA for ID 8
  126. Found audio description format PCMU for ID 0
  127. Found audio description format G726-32 for ID 2
  128. Found audio description format telephone-event for ID 101
  129. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  130. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  131. Peer audio RTP is at port 192.168.1.150:5010
  132. Looking for 06XXXXXXXX in SIP-PHONE-5171585574ee5e5a8ba8f6 (domain 192.168.1.200)
  133. list_route: hop: <sip:101@192.168.1.150:5060;transport=udp>
  134.  
  135. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  136. SIP/2.0 100 Trying
  137. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7b5cb758ecd1abce;received=192.168.1.150;rport=5060
  138. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  139. To: <sip:06XXXXXXXX@192.168.1.200>
  140. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  141. CSeq: 42847 INVITE
  142. Server: AskoziaPBX
  143. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  144. Supported: replaces, timer
  145. Contact: <sip:06XXXXXXXX@192.168.1.200:5060>
  146. Content-Length: 0
  147.  
  148.  
  149. <------------>
  150. Audio is at 5060
  151. Adding codec 0x4 (ulaw) to SDP
  152. Adding codec 0x8 (alaw) to SDP
  153. Adding codec 0x2 (gsm) to SDP
  154. Adding non-codec 0x1 (telephone-event) to SDP
  155. Reliably Transmitting (NAT) to 77.72.174.128:5060:
  156. INVITE sip:06XXXXXXXX@sip.easyvoip.com SIP/2.0
  157. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK4039100f;rport
  158. Max-Forwards: 70
  159. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  160. To: <sip:06XXXXXXXX@sip.easyvoip.com>
  161. Contact: <sip:USERNAME@10.192.26.204:5060>
  162. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  163. CSeq: 102 INVITE
  164. User-Agent: AskoziaPBX
  165. Date: Mon, 12 Dec 2011 11:44:39 GMT
  166. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  167. Supported: replaces, timer
  168. Content-Type: application/sdp
  169. Content-Length: 283
  170.  
  171. v=0
  172. o=root 576620492 576620492 IN IP4 10.192.26.204
  173. s=Asterisk PBX 1.8.4.4
  174. c=IN IP4 10.192.26.204
  175. t=0 0
  176. m=audio 10002 RTP/AVP 0 8 3 101
  177. a=rtpmap:0 PCMU/8000
  178. a=rtpmap:8 PCMA/8000
  179. a=rtpmap:3 GSM/8000
  180. a=rtpmap:101 telephone-event/8000
  181. a=fmtp:101 0-16
  182. a=ptime:20
  183. a=sendrecv
  184.  
  185. ---
  186. Really destroying SIP dialog '392a26e62d1be03119b61066759e8c52@192.168.1.2' Method: REGISTER
  187.  
  188. <--- SIP read from UDP:77.72.174.128:5060 --->
  189. SIP/2.0 401 Unauthorized
  190. Via: SIP/2.0/UDP 10.192.26.204:5060;rport;branch=z9hG4bK4039100f
  191. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  192. To: <sip:06XXXXXXXX@sip.easyvoip.com>
  193. Contact: sip:06XXXXXXXX@77.72.174.128:5060
  194. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  195. CSeq: 102 INVITE
  196. Server: (Very nice Sip Registrar/Proxy Server)
  197. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  198. WWW-Authenticate: Digest realm="sip.easyvoip.com",nonce="2334143046",algorithm=MD5
  199. Content-Length: 0
  200.  
  201. <------------->
  202. --- (11 headers 0 lines) ---
  203. Transmitting (NAT) to 77.72.174.128:5060:
  204. ACK sip:06XXXXXXXX@sip.easyvoip.com SIP/2.0
  205. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK4039100f;rport
  206. Max-Forwards: 70
  207. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  208. To: <sip:06XXXXXXXX@sip.easyvoip.com>
  209. Contact: <sip:USERNAME@10.192.26.204:5060>
  210. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  211. CSeq: 102 ACK
  212. User-Agent: AskoziaPBX
  213. Content-Length: 0
  214.  
  215.  
  216. ---
  217. Audio is at 5060
  218. Adding codec 0x4 (ulaw) to SDP
  219. Adding codec 0x8 (alaw) to SDP
  220. Adding codec 0x2 (gsm) to SDP
  221. Adding non-codec 0x1 (telephone-event) to SDP
  222. Reliably Transmitting (NAT) to 77.72.174.128:5060:
  223. INVITE sip:06XXXXXXXX@sip.easyvoip.com SIP/2.0
  224. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3c4c87d6;rport
  225. Max-Forwards: 70
  226. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  227. To: <sip:06XXXXXXXX@sip.easyvoip.com>
  228. Contact: <sip:USERNAME@10.192.26.204:5060>
  229. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  230. CSeq: 103 INVITE
  231. User-Agent: AskoziaPBX
  232. Authorization: Digest username="USERNAME", realm="sip.easyvoip.com", algorithm=MD5, uri="sip:06XXXXXXXX@sip.easyvoip.com", nonce="2334143046", response="37cf961ecd5d98dae15d6837757aa753"
  233. Date: Mon, 12 Dec 2011 11:44:40 GMT
  234. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  235. Supported: replaces, timer
  236. Content-Type: application/sdp
  237. Content-Length: 283
  238.  
  239. v=0
  240. o=root 576620492 576620493 IN IP4 10.192.26.204
  241. s=Asterisk PBX 1.8.4.4
  242. c=IN IP4 10.192.26.204
  243. t=0 0
  244. m=audio 10002 RTP/AVP 0 8 3 101
  245. a=rtpmap:0 PCMU/8000
  246. a=rtpmap:8 PCMA/8000
  247. a=rtpmap:3 GSM/8000
  248. a=rtpmap:101 telephone-event/8000
  249. a=fmtp:101 0-16
  250. a=ptime:20
  251. a=sendrecv
  252.  
  253. ---
  254.  
  255. <--- SIP read from UDP:77.72.174.128:5060 --->
  256. SIP/2.0 100 Trying
  257. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3c4c87d6;rport
  258. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  259. To: <sip:06XXXXXXXX@sip.easyvoip.com>
  260. Contact: sip:06XXXXXXXX@77.72.174.128:5060
  261. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  262. CSeq: 103 INVITE
  263. Server: (Very nice Sip Registrar/Proxy Server)
  264. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  265. Content-Length: 0
  266.  
  267. <------------->
  268. --- (10 headers 0 lines) ---
  269.  
  270. <--- SIP read from UDP:77.72.174.128:5060 --->
  271. SIP/2.0 183 Session progress
  272. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3c4c87d6;rport
  273. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  274. To: <sip:06XXXXXXXX@sip.easyvoip.com>;tag=300313ac4ed3c2d92cec5c
  275. Contact: sip:06XXXXXXXX@77.72.174.128:5060
  276. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  277. CSeq: 103 INVITE
  278. Server: (Very nice Sip Registrar/Proxy Server)
  279. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  280. Content-Type: application/sdp
  281. Content-Length: 200
  282.  
  283. v=0
  284. o=USERNAME 1323690280 1323690280 IN IP4 77.72.168.40
  285. s=SIP Call
  286. c=IN IP4 77.72.168.40
  287. t=0 0
  288. m=audio 24698 RTP/AVP 0 101
  289. a=rtpmap:0 PCMU/8000
  290. a=rtpmap:101 telephone-event/8000
  291. a=ptime:20
  292. <------------->
  293. --- (11 headers 9 lines) ---
  294. Found RTP audio format 0
  295. Found RTP audio format 101
  296. Found audio description format PCMU for ID 0
  297. Found audio description format telephone-event for ID 101
  298. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  299. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  300. Peer audio RTP is at port 77.72.168.40:24698
  301. Audio is at 5060
  302. Adding codec 0x4 (ulaw) to SDP
  303. Adding codec 0x8 (alaw) to SDP
  304. Adding non-codec 0x1 (telephone-event) to SDP
  305.  
  306. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  307. SIP/2.0 183 Session Progress
  308. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7b5cb758ecd1abce;received=192.168.1.150;rport=5060
  309. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  310. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as522f488c
  311. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  312. CSeq: 42847 INVITE
  313. Server: AskoziaPBX
  314. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  315. Supported: replaces, timer
  316. Contact: <sip:06XXXXXXXX@192.168.1.200:5060>
  317. Content-Type: application/sdp
  318. Content-Length: 260
  319.  
  320. v=0
  321. o=root 379387210 379387210 IN IP4 192.168.1.200
  322. s=Asterisk PBX 1.8.4.4
  323. c=IN IP4 192.168.1.200
  324. t=0 0
  325. m=audio 10196 RTP/AVP 0 8 101
  326. a=rtpmap:0 PCMU/8000
  327. a=rtpmap:8 PCMA/8000
  328. a=rtpmap:101 telephone-event/8000
  329. a=fmtp:101 0-16
  330. a=ptime:20
  331. a=sendrecv
  332.  
  333. <------------>
  334. Really destroying SIP dialog '2a25b55a29a64948559af39e1330ea8e@192.168.1.2' Method: REGISTER
  335.  
  336. <--- SIP read from UDP:77.72.174.128:5060 --->
  337. SIP/2.0 200 Ok
  338. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3c4c87d6;rport
  339. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  340. To: <sip:06XXXXXXXX@sip.easyvoip.com>;tag=300313ac4ed3c2d92cec5c
  341. Contact: sip:06XXXXXXXX@77.72.174.128:5060
  342. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  343. CSeq: 103 INVITE
  344. Server: (Very nice Sip Registrar/Proxy Server)
  345. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  346. Content-Type: application/sdp
  347. Content-Length: 200
  348.  
  349. v=0
  350. o=USERNAME 1323690294 1323690294 IN IP4 77.72.168.40
  351. s=SIP Call
  352. c=IN IP4 77.72.168.40
  353. t=0 0
  354. m=audio 24698 RTP/AVP 0 101
  355. a=rtpmap:0 PCMU/8000
  356. a=rtpmap:101 telephone-event/8000
  357. a=ptime:20
  358. <------------->
  359. --- (11 headers 9 lines) ---
  360. Found RTP audio format 0
  361. Found RTP audio format 101
  362. Found audio description format PCMU for ID 0
  363. Found audio description format telephone-event for ID 101
  364. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  365. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  366. Peer audio RTP is at port 77.72.168.40:24698
  367. list_route: hop: <sip:06XXXXXXXX@77.72.174.128:5060>
  368. set_destination: Parsing <sip:06XXXXXXXX@77.72.174.128:5060> for address/port to send to
  369. set_destination: set destination to 77.72.174.128:5060
  370. Transmitting (NAT) to 77.72.174.128:5060:
  371. ACK sip:06XXXXXXXX@77.72.174.128:5060 SIP/2.0
  372. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK32cf4c25;rport
  373. Max-Forwards: 70
  374. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  375. To: <sip:06XXXXXXXX@sip.easyvoip.com>;tag=300313ac4ed3c2d92cec5c
  376. Contact: <sip:USERNAME@10.192.26.204:5060>
  377. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  378. CSeq: 103 ACK
  379. User-Agent: AskoziaPBX
  380. Content-Length: 0
  381.  
  382.  
  383. ---
  384. Audio is at 5060
  385. Adding codec 0x4 (ulaw) to SDP
  386. Adding codec 0x8 (alaw) to SDP
  387. Adding non-codec 0x1 (telephone-event) to SDP
  388.  
  389. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  390. SIP/2.0 200 OK
  391. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7b5cb758ecd1abce;received=192.168.1.150;rport=5060
  392. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  393. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as522f488c
  394. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  395. CSeq: 42847 INVITE
  396. Server: AskoziaPBX
  397. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  398. Supported: replaces, timer
  399. Contact: <sip:06XXXXXXXX@192.168.1.200:5060>
  400. Content-Type: application/sdp
  401. Content-Length: 260
  402.  
  403. v=0
  404. o=root 379387210 379387211 IN IP4 192.168.1.200
  405. s=Asterisk PBX 1.8.4.4
  406. c=IN IP4 192.168.1.200
  407. t=0 0
  408. m=audio 10196 RTP/AVP 0 8 101
  409. a=rtpmap:0 PCMU/8000
  410. a=rtpmap:8 PCMA/8000
  411. a=rtpmap:101 telephone-event/8000
  412. a=fmtp:101 0-16
  413. a=ptime:20
  414. a=sendrecv
  415.  
  416. <------------>
  417.  
  418. <--- SIP read from UDP:192.168.1.150:5060 --->
  419. ACK sip:06XXXXXXXX@192.168.1.200:5060 SIP/2.0
  420. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK1dea0f6b71d559a1
  421. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  422. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as522f488c
  423. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  424. Supported: path
  425. X-Grandstream-PBX: true
  426. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="1ab59689", response="e9789d0f23f0c051e85b399e2867409d"
  427. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  428. CSeq: 42847 ACK
  429. User-Agent: Grandstream GXP1200 1.2.5.3
  430. Max-Forwards: 70
  431. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  432. Content-Length: 0
  433.  
  434. <------------->
  435. --- (14 headers 0 lines) ---
  436.  
  437. <--- SIP read from UDP:192.168.1.150:5060 --->
  438. BYE sip:06XXXXXXXX@192.168.1.200:5060 SIP/2.0
  439. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK2841e3d5eb0d6219
  440. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  441. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as522f488c
  442. Supported: path
  443. X-Grandstream-PBX: true
  444. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200:5060", nonce="1ab59689", response="f345e8cf8f05fad55cf8deb3847f094e"
  445. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  446. CSeq: 42848 BYE
  447. User-Agent: Grandstream GXP1200 1.2.5.3
  448. Max-Forwards: 70
  449. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  450. Reason: SIP ;text="Onhook event"
  451. Content-Length: 0
  452.  
  453. <------------->
  454. --- (14 headers 0 lines) ---
  455. Sending to 192.168.1.150:5060 (NAT)
  456. Scheduling destruction of SIP dialog '95a53eef3afeb0c2@192.168.1.150' in 6400 ms (Method: BYE)
  457.  
  458. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  459. SIP/2.0 200 OK
  460. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK2841e3d5eb0d6219;received=192.168.1.150;rport=5060
  461. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=8d6540afad7801d9
  462. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as522f488c
  463. Call-ID: 95a53eef3afeb0c2@192.168.1.150
  464. CSeq: 42848 BYE
  465. Server: AskoziaPBX
  466. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  467. Supported: replaces, timer
  468. Content-Length: 0
  469.  
  470.  
  471. <------------>
  472. Scheduling destruction of SIP dialog '290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com' in 79296 ms (Method: INVITE)
  473. set_destination: Parsing <sip:06XXXXXXXX@77.72.174.128:5060> for address/port to send to
  474. set_destination: set destination to 77.72.174.128:5060
  475. Reliably Transmitting (NAT) to 77.72.174.128:5060:
  476. BYE sip:06XXXXXXXX@77.72.174.128:5060 SIP/2.0
  477. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK26b4f17a;rport
  478. Max-Forwards: 70
  479. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  480. To: <sip:06XXXXXXXX@sip.easyvoip.com>;tag=300313ac4ed3c2d92cec5c
  481. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  482. CSeq: 104 BYE
  483. User-Agent: AskoziaPBX
  484. Authorization: Digest username="USERNAME", realm="sip.easyvoip.com", algorithm=MD5, uri="sip:06XXXXXXXX@77.72.174.128:5060", nonce="2334143046", response="94fc6f8762be2d3abd4f005dbef2c6a6"
  485. X-Asterisk-HangupCause: Normal Clearing
  486. X-Asterisk-HangupCauseCode: 16
  487. Content-Length: 0
  488.  
  489.  
  490. ---
  491.  
  492. <--- SIP read from UDP:77.72.174.128:5060 --->
  493. SIP/2.0 200 Ok
  494. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK26b4f17a;rport
  495. From: "Default Extension" <sip:USERNAME@sip.easyvoip.com>;tag=as7e82875d
  496. To: <sip:06XXXXXXXX@sip.easyvoip.com>;tag=300313ac4ed3c2d92cec5c
  497. Contact: sip:06XXXXXXXX@77.72.174.128:5060
  498. Call-ID: 290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com
  499. CSeq: 104 BYE
  500. Server: (Very nice Sip Registrar/Proxy Server)
  501. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  502. Content-Length: 0
  503.  
  504. <------------->
  505. --- (10 headers 0 lines) ---
  506. Really destroying SIP dialog '290e956b1ec49b3c581df49e110a2eff@sip.easyvoip.com' Method: INVITE
  507. Really destroying SIP dialog '95a53eef3afeb0c2@192.168.1.150' Method: BYE
  508. Reliably Transmitting (NAT) to 192.168.1.150:5060:
  509. OPTIONS sip:101@192.168.1.150:5060;transport=udp SIP/2.0
  510. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK6d5726d0;rport
  511. Max-Forwards: 70
  512. From: "asterisk" <sip:asterisk@192.168.1.200>;tag=as2be806e5
  513. To: <sip:101@192.168.1.150:5060;transport=udp>
  514. Contact: <sip:asterisk@192.168.1.200:5060>
  515. Call-ID: 4318fff34d9c76013d21c487138fa518@192.168.1.200:5060
  516. CSeq: 102 OPTIONS
  517. User-Agent: AskoziaPBX
  518. Date: Mon, 12 Dec 2011 11:45:10 GMT
  519. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  520. Supported: replaces, timer
  521. Content-Length: 0
  522.  
  523.  
  524. ---
  525.  
  526. <--- SIP read from UDP:192.168.1.150:5060 --->
  527. SIP/2.0 200 OK
  528. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK6d5726d0;rport
  529. From: "asterisk" <sip:asterisk@192.168.1.200>;tag=as2be806e5
  530. To: <sip:101@192.168.1.150:5060;transport=udp>;tag=54d780c74dedb8f6
  531. Call-ID: 4318fff34d9c76013d21c487138fa518@192.168.1.200:5060
  532. CSeq: 102 OPTIONS
  533. User-Agent: Grandstream GXP1200 1.2.5.3
  534. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  535. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  536. Supported: replaces, timer
  537. Content-Length: 0
  538.  
  539. <------------->
  540. --- (11 headers 0 lines) ---
  541. Really destroying SIP dialog '4318fff34d9c76013d21c487138fa518@192.168.1.200:5060' Method: OPTIONS
  542. Reliably Transmitting (NAT) to 77.72.174.128:5060:
  543. OPTIONS sip:sip.easyvoip.com SIP/2.0
  544. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK275dcf8e;rport
  545. Max-Forwards: 70
  546. From: "asterisk" <sip:asterisk@10.192.26.204>;tag=as79455649
  547. To: <sip:sip.easyvoip.com>
  548. Contact: <sip:asterisk@10.192.26.204:5060>
  549. Call-ID: 3af915b815d5df161b833c4c72928903@10.192.26.204:5060
  550. CSeq: 102 OPTIONS
  551. User-Agent: AskoziaPBX
  552. Date: Mon, 12 Dec 2011 11:45:11 GMT
  553. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  554. Supported: replaces, timer
  555. Content-Length: 0
  556.  
  557.  
  558. ---
  559.  
  560. <--- SIP read from UDP:77.72.174.128:5060 --->
  561. SIP/2.0 200 Ok
  562. Via: SIP/2.0/UDP 10.192.26.204:5060;rport;branch=z9hG4bK275dcf8e
  563. From: "asterisk" <sip:asterisk@10.192.26.204>;tag=as79455649
  564. To: <sip:sip.easyvoip.com>
  565. Contact: sip:77.72.174.128:5060
  566. Call-ID: 3af915b815d5df161b833c4c72928903@10.192.26.204:5060
  567. CSeq: 102 OPTIONS
  568. Supported: foo
  569. User-Agent: (Very nice Sip Registrar/Proxy Server)
  570. Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
  571. Accept: application/sdp
  572. Content-Length: 0
  573.  
  574. <------------->
  575. --- (12 headers 0 lines) ---
  576. Really destroying SIP dialog '3af915b815d5df161b833c4c72928903@10.192.26.204:5060' Method: OPTIONS
  577.  
  578.  
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