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  1.  
  2. INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
  4. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  5. To: <sip:06XXXXXXXX@192.168.1.200>
  6. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  7. Supported: replaces, timer, path
  8. X-Grandstream-PBX: true
  9. P-Early-Media: Supported
  10. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  11. CSeq: 33404 INVITE
  12. User-Agent: Grandstream GXP1200 1.2.5.3
  13. Max-Forwards: 70
  14. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  15. Content-Type: application/sdp
  16. Content-Length: 313
  17.  
  18. v=0
  19. o=101 8000 8000 IN IP4 192.168.1.150
  20. s=SIP Call
  21. c=IN IP4 192.168.1.150
  22. t=0 0
  23. m=audio 5014 RTP/AVP 9 18 8 0 2 101
  24. a=sendrecv
  25. a=rtpmap:9 G722/8000
  26. a=rtpmap:18 G729/8000
  27. a=rtpmap:8 PCMA/8000
  28. a=rtpmap:0 PCMU/8000
  29. a=rtpmap:2 G726-32/8000
  30. a=ptime:20
  31. a=rtpmap:101 telephone-event/8000
  32. a=fmtp:101 0-11
  33. <------------->
  34. --- (15 headers 15 lines) ---
  35. Sending to 192.168.1.150:5060 (no NAT)
  36. Using INVITE request as basis request - 1f25e87efe2b6f84@192.168.1.150
  37. Found peer '101' for '101' from 192.168.1.150:5060
  38.  
  39. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  40. SIP/2.0 401 Unauthorized
  41. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02;received=192.168.1.150;rport=5060
  42. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  43. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as4c3d30a9
  44. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  45. CSeq: 33404 INVITE
  46. Server: AskoziaPBX
  47. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  48. Supported: replaces, timer
  49. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75a6cfa2"
  50. Content-Length: 0
  51.  
  52.  
  53. <------------>
  54. Scheduling destruction of SIP dialog '1f25e87efe2b6f84@192.168.1.150' in 6400 ms (Method: INVITE)
  55.  
  56. <--- SIP read from UDP:192.168.1.150:5060 --->
  57. ACK sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  58. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
  59. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  60. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as4c3d30a9
  61. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  62. Supported: path
  63. X-Grandstream-PBX: true
  64. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  65. CSeq: 33404 ACK
  66. User-Agent: Grandstream GXP1200 1.2.5.3
  67. Max-Forwards: 70
  68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  69. Content-Length: 0
  70.  
  71. <------------->
  72. --- (13 headers 0 lines) ---
  73.  
  74. <--- SIP read from UDP:192.168.1.150:5060 --->
  75. INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  76. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  77. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  78. To: <sip:06XXXXXXXX@192.168.1.200>
  79. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  80. Supported: replaces, timer, path
  81. X-Grandstream-PBX: true
  82. P-Early-Media: Supported
  83. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
  84. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  85. CSeq: 33405 INVITE
  86. User-Agent: Grandstream GXP1200 1.2.5.3
  87. Max-Forwards: 70
  88. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  89. Content-Type: application/sdp
  90. Content-Length: 313
  91.  
  92. v=0
  93. o=101 8000 8001 IN IP4 192.168.1.150
  94. s=SIP Call
  95. c=IN IP4 192.168.1.150
  96. t=0 0
  97. m=audio 5014 RTP/AVP 9 18 8 0 2 101
  98. a=sendrecv
  99. a=rtpmap:9 G722/8000
  100. a=rtpmap:18 G729/8000
  101. a=rtpmap:8 PCMA/8000
  102. a=rtpmap:0 PCMU/8000
  103. a=rtpmap:2 G726-32/8000
  104. a=ptime:20
  105. a=rtpmap:101 telephone-event/8000
  106. a=fmtp:101 0-11
  107. <------------->
  108. --- (16 headers 15 lines) ---
  109. Sending to 192.168.1.150:5060 (NAT)
  110. Using INVITE request as basis request - 1f25e87efe2b6f84@192.168.1.150
  111. Found peer '101' for '101' from 192.168.1.150:5060
  112. Found RTP audio format 9
  113. Found RTP audio format 18
  114. Found RTP audio format 8
  115. Found RTP audio format 0
  116. Found RTP audio format 2
  117. Found RTP audio format 101
  118. Found audio description format G722 for ID 9
  119. Found audio description format G729 for ID 18
  120. Found audio description format PCMA for ID 8
  121. Found audio description format PCMU for ID 0
  122. Found audio description format G726-32 for ID 2
  123. Found audio description format telephone-event for ID 101
  124. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  125. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  126. Peer audio RTP is at port 192.168.1.150:5014
  127. Looking for 06XXXXXXXX in SIP-PHONE-5171585574ee5e5a8ba8f6 (domain 192.168.1.200)
  128. list_route: hop: <sip:101@192.168.1.150:5060;transport=udp>
  129.  
  130. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  131. SIP/2.0 100 Trying
  132. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  133. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  134. To: <sip:06XXXXXXXX@192.168.1.200>
  135. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  136. CSeq: 33405 INVITE
  137. Server: AskoziaPBX
  138. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  139. Supported: replaces, timer
  140. Contact: <sip:06XXXXXXXX@192.168.1.200:5060>
  141. Content-Length: 0
  142.  
  143.  
  144. <------------>
  145. Audio is at 5060
  146. Adding codec 0x4 (ulaw) to SDP
  147. Adding codec 0x8 (alaw) to SDP
  148. Adding codec 0x2 (gsm) to SDP
  149. Adding non-codec 0x1 (telephone-event) to SDP
  150. Reliably Transmitting (NAT) to 91.121.129.17:5060:
  151. INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  152. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  153. Max-Forwards: 70
  154. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  155. To: <sip:06XXXXXXXX@sip.ovh.net>
  156. Contact: <sip:USERNAME@10.192.26.204:5060>
  157. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  158. CSeq: 102 INVITE
  159. User-Agent: AskoziaPBX
  160. Date: Mon, 12 Dec 2011 11:48:06 GMT
  161. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  162. Supported: replaces, timer
  163. Content-Type: application/sdp
  164. Content-Length: 285
  165.  
  166. v=0
  167. o=root 1595019147 1595019147 IN IP4 10.192.26.204
  168. s=Asterisk PBX 1.8.4.4
  169. c=IN IP4 10.192.26.204
  170. t=0 0
  171. m=audio 10072 RTP/AVP 0 8 3 101
  172. a=rtpmap:0 PCMU/8000
  173. a=rtpmap:8 PCMA/8000
  174. a=rtpmap:3 GSM/8000
  175. a=rtpmap:101 telephone-event/8000
  176. a=fmtp:101 0-16
  177. a=ptime:20
  178. a=sendrecv
  179.  
  180. ---
  181. Retransmitting #1 (NAT) to 91.121.129.17:5060:
  182. INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  183. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  184. Max-Forwards: 70
  185. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  186. To: <sip:06XXXXXXXX@sip.ovh.net>
  187. Contact: <sip:USERNAME@10.192.26.204:5060>
  188. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  189. CSeq: 102 INVITE
  190. User-Agent: AskoziaPBX
  191. Date: Mon, 12 Dec 2011 11:48:06 GMT
  192. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  193. Supported: replaces, timer
  194. Content-Type: application/sdp
  195. Content-Length: 285
  196.  
  197. v=0
  198. o=root 1595019147 1595019147 IN IP4 10.192.26.204
  199. s=Asterisk PBX 1.8.4.4
  200. c=IN IP4 10.192.26.204
  201. t=0 0
  202. m=audio 10072 RTP/AVP 0 8 3 101
  203. a=rtpmap:0 PCMU/8000
  204. a=rtpmap:8 PCMA/8000
  205. a=rtpmap:3 GSM/8000
  206. a=rtpmap:101 telephone-event/8000
  207. a=fmtp:101 0-16
  208. a=ptime:20
  209. a=sendrecv
  210.  
  211. ---
  212.  
  213. <--- SIP read from UDP:91.121.129.17:5060 --->
  214. SIP/2.0 407 authentication required
  215. Allow: UPDATE,REFER,INFO
  216. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  217. Contact: <sip:06XXXXXXXX@91.121.129.17:5060;user=phone>
  218. CSeq: 102 INVITE
  219. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  220. Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
  221. Server: Cirpack/v4.42j (gw_sip)
  222. To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
  223. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
  224. Content-Length: 0
  225.  
  226. <------------->
  227. --- (11 headers 0 lines) ---
  228. Transmitting (NAT) to 91.121.129.17:5060:
  229. ACK sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  230. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  231. Max-Forwards: 70
  232. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  233. To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
  234. Contact: <sip:USERNAME@10.192.26.204:5060>
  235. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  236. CSeq: 102 ACK
  237. User-Agent: AskoziaPBX
  238. Content-Length: 0
  239.  
  240.  
  241. ---
  242. Audio is at 5060
  243. Adding codec 0x4 (ulaw) to SDP
  244. Adding codec 0x8 (alaw) to SDP
  245. Adding codec 0x2 (gsm) to SDP
  246. Adding non-codec 0x1 (telephone-event) to SDP
  247. Reliably Transmitting (NAT) to 91.121.129.17:5060:
  248. INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  249. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  250. Max-Forwards: 70
  251. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  252. To: <sip:06XXXXXXXX@sip.ovh.net>
  253. Contact: <sip:USERNAME@10.192.26.204:5060>
  254. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  255. CSeq: 103 INVITE
  256. User-Agent: AskoziaPBX
  257. Proxy-Authorization: Digest username="USERNAME", realm="sip.ovh.net", algorithm=MD5, uri="sip:06XXXXXXXX@sip.ovh.net", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
  258. Date: Mon, 12 Dec 2011 11:48:07 GMT
  259. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  260. Supported: replaces, timer
  261. Content-Type: application/sdp
  262. Content-Length: 285
  263.  
  264. v=0
  265. o=root 1595019147 1595019148 IN IP4 10.192.26.204
  266. s=Asterisk PBX 1.8.4.4
  267. c=IN IP4 10.192.26.204
  268. t=0 0
  269. m=audio 10072 RTP/AVP 0 8 3 101
  270. a=rtpmap:0 PCMU/8000
  271. a=rtpmap:8 PCMA/8000
  272. a=rtpmap:3 GSM/8000
  273. a=rtpmap:101 telephone-event/8000
  274. a=fmtp:101 0-16
  275. a=ptime:20
  276. a=sendrecv
  277.  
  278. ---
  279.  
  280. <--- SIP read from UDP:91.121.129.17:5060 --->
  281. SIP/2.0 407 authentication required
  282. Allow: UPDATE,REFER,INFO
  283. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  284. Contact: <sip:06XXXXXXXX@91.121.129.17:5060;user=phone>
  285. CSeq: 102 INVITE
  286. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  287. Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
  288. Server: Cirpack/v4.42j (gw_sip)
  289. To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
  290. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
  291. Content-Length: 0
  292.  
  293. <------------->
  294. --- (11 headers 0 lines) ---
  295. Transmitting (NAT) to 91.121.129.17:5060:
  296. ACK sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  297. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  298. Max-Forwards: 70
  299. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  300. To: <sip:06XXXXXXXX@sip.ovh.net>
  301. Contact: <sip:USERNAME@10.192.26.204:5060>
  302. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  303. CSeq: 102 ACK
  304. User-Agent: AskoziaPBX
  305. Content-Length: 0
  306.  
  307.  
  308. ---
  309. Retransmitting #1 (NAT) to 91.121.129.17:5060:
  310. INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
  311. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  312. Max-Forwards: 70
  313. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  314. To: <sip:06XXXXXXXX@sip.ovh.net>
  315. Contact: <sip:USERNAME@10.192.26.204:5060>
  316. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  317. CSeq: 103 INVITE
  318. User-Agent: AskoziaPBX
  319. Proxy-Authorization: Digest username="USERNAME", realm="sip.ovh.net", algorithm=MD5, uri="sip:06XXXXXXXX@sip.ovh.net", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
  320. Date: Mon, 12 Dec 2011 11:48:07 GMT
  321. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  322. Supported: replaces, timer
  323. Content-Type: application/sdp
  324. Content-Length: 285
  325.  
  326. v=0
  327. o=root 1595019147 1595019148 IN IP4 10.192.26.204
  328. s=Asterisk PBX 1.8.4.4
  329. c=IN IP4 10.192.26.204
  330. t=0 0
  331. m=audio 10072 RTP/AVP 0 8 3 101
  332. a=rtpmap:0 PCMU/8000
  333. a=rtpmap:8 PCMA/8000
  334. a=rtpmap:3 GSM/8000
  335. a=rtpmap:101 telephone-event/8000
  336. a=fmtp:101 0-16
  337. a=ptime:20
  338. a=sendrecv
  339.  
  340. ---
  341.  
  342. <--- SIP read from UDP:91.121.129.17:5060 --->
  343. SIP/2.0 100 Trying
  344. Allow: UPDATE,REFER,INFO
  345. Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
  346. Contact: <sip:91.121.129.17:5060>
  347. CSeq: 103 INVITE
  348. From: "Default Extension" <sip:USERNAME@sip.ovh.net>;tag=as4523e5ab
  349. Server: Cirpack/v4.42j (gw_sip)
  350. To: <sip:06XXXXXXXX@sip.ovh.net>
  351. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK0b23a06a
  352. Content-Length: 0
  353.  
  354. <------------->
  355. --- (10 headers 0 lines) ---
  356.  
  357. <--- SIP read from UDP:192.168.1.150:5060 --->
  358. CANCEL sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  359. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  360. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  361. To: <sip:06XXXXXXXX@192.168.1.200>
  362. Supported: path
  363. X-Grandstream-PBX: true
  364. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  365. CSeq: 33405 CANCEL
  366. User-Agent: Grandstream GXP1200 1.2.5.3
  367. Max-Forwards: 70
  368. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  369. Content-Length: 0
  370.  
  371. <------------->
  372. --- (12 headers 0 lines) ---
  373. Sending to 192.168.1.150:5060 (NAT)
  374.  
  375. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  376. SIP/2.0 487 Request Terminated
  377. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  378. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  379. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
  380. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  381. CSeq: 33405 INVITE
  382. Server: AskoziaPBX
  383. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  384. Supported: replaces, timer
  385. Content-Length: 0
  386.  
  387.  
  388. <------------>
  389.  
  390. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  391. SIP/2.0 200 OK
  392. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  393. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  394. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
  395. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  396. CSeq: 33405 CANCEL
  397. Server: AskoziaPBX
  398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  399. Supported: replaces, timer
  400. Content-Length: 0
  401.  
  402.  
  403. <------------>
  404. Scheduling destruction of SIP dialog '75f7814960e8c0081359b3700cc6f332@sip.ovh.net' in 53440 ms (Method: INVITE)
  405. Retransmitting #1 (NAT) to 192.168.1.150:5060:
  406. SIP/2.0 487 Request Terminated
  407. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  408. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  409. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
  410. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  411. CSeq: 33405 INVITE
  412. Server: AskoziaPBX
  413. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  414. Supported: replaces, timer
  415. Content-Length: 0
  416.  
  417.  
  418. ---
  419.  
  420. <--- SIP read from UDP:192.168.1.150:5060 --->
  421. ACK sip:06XXXXXXXX@192.168.1.200 SIP/2.0
  422. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  423. From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
  424. To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
  425. Contact: <sip:101@192.168.1.150:5060;transport=udp>
  426. Supported: path
  427. X-Grandstream-PBX: true
  428. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
  429. Call-ID: 1f25e87efe2b6f84@192.168.1.150
  430. CSeq: 33405 ACK
  431. User-Agent: Grandstream GXP1200 1.2.5.3
  432. Max-Forwards: 70
  433. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  434. Content-Length: 0
  435.  
  436. <------------->
  437. --- (14 headers 0 lines) ---
  438. Really destroying SIP dialog '1f25e87efe2b6f84@192.168.1.150' Method: ACK
  439. Really destroying SIP dialog '2a25b55a29a64948559af39e1330ea8e@192.168.1.2' Method: REGISTER
  440.  
  441.  
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