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  1. U 192.168.2.11:5060 -> 192.168.2.102:5060
  2. INVITE sip:16034531989@opensips1.test.com;user=phone SIP/2.0.
  3. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK853f84ceEA527609.
  4. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  5. To: <sip:16034531989@opensips1.test.com;user=phone>.
  6. CSeq: 1 INVITE.
  7. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  8. Contact: <sip:1001@192.168.2.11>.
  9. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
  10. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
  11. Supported: 100rel,replaces.
  12. Allow-Events: talk,hold,conference.
  13. Max-Forwards: 70.
  14. Content-Type: application/sdp.
  15. Content-Length: 250.
  16. .
  17. v=0.
  18. o=- 1324311626 1324311626 IN IP4 192.168.2.11.
  19. s=Polycom IP Phone.
  20. c=IN IP4 192.168.2.11.
  21. t=0 0.
  22. a=sendrecv.
  23. m=audio 10006 RTP/AVP 0 8 18 101.
  24. a=rtpmap:0 PCMU/8000.
  25. a=rtpmap:8 PCMA/8000.
  26. a=rtpmap:18 G729/8000.
  27. a=rtpmap:101 telephone-event/8000.
  28.  
  29.  
  30. U 192.168.2.102:5060 -> 192.168.2.11:5060
  31. SIP/2.0 407 Proxy Authentication Required.
  32. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK853f84ceEA527609.
  33. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  34. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=4899a85fdda7a45fc4d7b6eb4e737879.4ee5.
  35. CSeq: 1 INVITE.
  36. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  37. Proxy-Authenticate: Digest realm="opensips1.test.com", nonce="4eef645a00000003293e36ae452e5356403d199ba09813a4".
  38. opensips1.test.com.
  39. Content-Length: 0.
  40. Warning: 392 192.168.2.102:5060 "Noisy feedback tells: pid=1307 req_src_ip=192.168.2.11 req_src_port=5060 in_uri=sip:16034531989@opensips1.test.com;user=phone out_uri=sip:16034531989@opensips1.test.com;user=phone via_cnt==1".
  41. .
  42.  
  43.  
  44. U 192.168.2.111:5060 -> 192.168.2.102:5060
  45. SIP/2.0 200 OK.
  46. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bK3d32.a3ba3543.0;received=192.168.2.102.
  47. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK3df8b58b6DF9536.
  48. Record-Route: <sip:opensips1.test.com;lr>.
  49. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=B5A87FF0-81E03165.
  50. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as2fd05f4c.
  51. Call-ID: 1bab8224-9c5f96ea-701a0d7f@192.168.2.11.
  52. CSeq: 2 INVITE.
  53. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  54. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  55. Supported: replaces, timer.
  56. Contact: <sip:16034531989@192.168.2.111:5060>.
  57. Content-Type: application/sdp.
  58. Content-Length: 313.
  59. .
  60. v=0.
  61. o=root 331845838 331845839 IN IP4 192.168.2.111.
  62. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  63. c=IN IP4 192.168.2.111.
  64. t=0 0.
  65. m=audio 31576 RTP/AVP 8 0 101.
  66. a=rtpmap:8 PCMA/8000.
  67. a=rtpmap:0 PCMU/8000.
  68. a=rtpmap:101 telephone-event/8000.
  69. a=fmtp:101 0-16.
  70. a=silenceSupp:off - - - -.
  71. a=ptime:20.
  72. a=sendrecv.
  73.  
  74.  
  75. U 192.168.2.11:5060 -> 192.168.2.102:5060
  76. ACK sip:16034531989@opensips1.test.com SIP/2.0.
  77. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK853f84ceEA527609.
  78. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  79. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=4899a85fdda7a45fc4d7b6eb4e737879.4ee5.
  80. CSeq: 1 ACK.
  81. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  82. Contact: <sip:1001@192.168.2.11>.
  83. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
  84. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
  85. Max-Forwards: 70.
  86. Content-Length: 0.
  87. .
  88.  
  89.  
  90. U 192.168.2.11:5060 -> 192.168.2.102:5060
  91. INVITE sip:16034531989@opensips1.test.com;user=phone SIP/2.0.
  92. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  93. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  94. To: <sip:16034531989@opensips1.test.com;user=phone>.
  95. CSeq: 2 INVITE.
  96. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  97. Contact: <sip:1001@192.168.2.11>.
  98. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
  99. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
  100. Supported: 100rel,replaces.
  101. Allow-Events: talk,hold,conference.
  102. Proxy-Authorization: Digest username="1001", realm="opensips1.test.com", nonce="4eef645a00000003293e36ae452e5356403d199ba09813a4", uri="sip:16034531989@opensips1.test.com;user=phone", response="4fc5718e13822efcb9264a3b9bd242b8", algorithm=MD5.
  103. Max-Forwards: 70.
  104. Content-Type: application/sdp.
  105. Content-Length: 250.
  106. .
  107. v=0.
  108. o=- 1324311626 1324311626 IN IP4 192.168.2.11.
  109. s=Polycom IP Phone.
  110. c=IN IP4 192.168.2.11.
  111. t=0 0.
  112. a=sendrecv.
  113. m=audio 10006 RTP/AVP 0 8 18 101.
  114. a=rtpmap:0 PCMU/8000.
  115. a=rtpmap:8 PCMA/8000.
  116. a=rtpmap:18 G729/8000.
  117. a=rtpmap:101 telephone-event/8000.
  118.  
  119.  
  120. U 192.168.2.102:5060 -> 192.168.2.11:5060
  121. SIP/2.0 100 Giving a try.
  122. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  123. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  124. To: <sip:16034531989@opensips1.test.com;user=phone>.
  125. CSeq: 2 INVITE.
  126. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  127. opensips1.test.com.
  128. Content-Length: 0.
  129. Warning: 392 192.168.2.102:5060 "Noisy feedback tells: pid=1310 req_src_ip=192.168.2.11 req_src_port=5060 in_uri=sip:16034531989@opensips1.test.com;user=phone out_uri=sip:16034531989@astcluster.test.com:5060;user=phone via_cnt==1".
  130. .
  131.  
  132.  
  133. U 192.168.2.102:5060 -> 192.168.2.6:5060
  134. INVITE sip:16034531989@astcluster.test.com:5060;user=phone SIP/2.0.
  135. Record-Route: <sip:opensips1.test.com;lr>.
  136. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0.
  137. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  138. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  139. To: <sip:16034531989@opensips1.test.com;user=phone>.
  140. CSeq: 2 INVITE.
  141. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  142. Contact: <sip:1001@192.168.2.11>.
  143. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
  144. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
  145. Supported: 100rel,replaces.
  146. Allow-Events: talk,hold,conference.
  147. Proxy-Authorization: Digest username="1001", realm="opensips1.test.com", nonce="4eef645a00000003293e36ae452e5356403d199ba09813a4", uri="sip:16034531989@opensips1.test.com;user=phone", response="4fc5718e13822efcb9264a3b9bd242b8", algorithm=MD5.
  148. Max-Forwards: 70.
  149. Content-Type: application/sdp.
  150. Content-Length: 250.
  151. .
  152. v=0.
  153. o=- 1324311626 1324311626 IN IP4 192.168.2.11.
  154. s=Polycom IP Phone.
  155. c=IN IP4 192.168.2.11.
  156. t=0 0.
  157. a=sendrecv.
  158. m=audio 10006 RTP/AVP 0 8 18 101.
  159. a=rtpmap:0 PCMU/8000.
  160. a=rtpmap:8 PCMA/8000.
  161. a=rtpmap:18 G729/8000.
  162. a=rtpmap:101 telephone-event/8000.
  163.  
  164.  
  165. U 192.168.2.111:5060 -> 192.168.2.102:5060
  166. SIP/2.0 100 Trying.
  167. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  168. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  169. Record-Route: <sip:opensips1.test.com;lr>.
  170. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  171. To: <sip:16034531989@opensips1.test.com;user=phone>.
  172. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  173. CSeq: 2 INVITE.
  174. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  175. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  176. Supported: replaces, timer.
  177. Contact: <sip:16034531989@192.168.2.111:5060>.
  178. Content-Length: 0.
  179. .
  180.  
  181.  
  182. U 192.168.2.102:5060 -> 192.168.2.6:5060
  183. INVITE sip:16034531989@astcluster.test.com:5060;user=phone SIP/2.0.
  184. Record-Route: <sip:opensips1.test.com;lr>.
  185. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0.
  186. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  187. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  188. To: <sip:16034531989@opensips1.test.com;user=phone>.
  189. CSeq: 2 INVITE.
  190. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  191. Contact: <sip:1001@192.168.2.11>.
  192. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
  193. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
  194. Supported: 100rel,replaces.
  195. Allow-Events: talk,hold,conference.
  196. Proxy-Authorization: Digest username="1001", realm="opensips1.test.com", nonce="4eef645a00000003293e36ae452e5356403d199ba09813a4", uri="sip:16034531989@opensips1.test.com;user=phone", response="4fc5718e13822efcb9264a3b9bd242b8", algorithm=MD5.
  197. Max-Forwards: 70.
  198. Content-Type: application/sdp.
  199. Content-Length: 250.
  200. .
  201. v=0.
  202. o=- 1324311626 1324311626 IN IP4 192.168.2.11.
  203. s=Polycom IP Phone.
  204. c=IN IP4 192.168.2.11.
  205. t=0 0.
  206. a=sendrecv.
  207. m=audio 10006 RTP/AVP 0 8 18 101.
  208. a=rtpmap:0 PCMU/8000.
  209. a=rtpmap:8 PCMA/8000.
  210. a=rtpmap:18 G729/8000.
  211. a=rtpmap:101 telephone-event/8000.
  212.  
  213.  
  214. U 192.168.2.111:5060 -> 192.168.2.102:5060
  215. SIP/2.0 100 Trying.
  216. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  217. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  218. Record-Route: <sip:opensips1.test.com;lr>.
  219. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  220. To: <sip:16034531989@opensips1.test.com;user=phone>.
  221. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  222. CSeq: 2 INVITE.
  223. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  224. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  225. Supported: replaces, timer.
  226. Contact: <sip:16034531989@192.168.2.111:5060>.
  227. Content-Length: 0.
  228. .
  229.  
  230.  
  231. U 192.168.2.111:5060 -> 192.168.2.102:5060
  232. SIP/2.0 200 OK.
  233. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bK3d32.a3ba3543.0;received=192.168.2.102.
  234. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK3df8b58b6DF9536.
  235. Record-Route: <sip:opensips1.test.com;lr>.
  236. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=B5A87FF0-81E03165.
  237. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as2fd05f4c.
  238. Call-ID: 1bab8224-9c5f96ea-701a0d7f@192.168.2.11.
  239. CSeq: 2 INVITE.
  240. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  242. Supported: replaces, timer.
  243. Contact: <sip:16034531989@192.168.2.111:5060>.
  244. Content-Type: application/sdp.
  245. Content-Length: 313.
  246. .
  247. v=0.
  248. o=root 331845838 331845839 IN IP4 192.168.2.111.
  249. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  250. c=IN IP4 192.168.2.111.
  251. t=0 0.
  252. m=audio 31576 RTP/AVP 8 0 101.
  253. a=rtpmap:8 PCMA/8000.
  254. a=rtpmap:0 PCMU/8000.
  255. a=rtpmap:101 telephone-event/8000.
  256. a=fmtp:101 0-16.
  257. a=silenceSupp:off - - - -.
  258. a=ptime:20.
  259. a=sendrecv.
  260.  
  261.  
  262. U 192.168.2.111:5060 -> 192.168.2.102:5060
  263. SIP/2.0 200 OK.
  264. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  265. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  266. Record-Route: <sip:opensips1.test.com;lr>.
  267. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  268. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  269. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  270. CSeq: 2 INVITE.
  271. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  272. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  273. Supported: replaces, timer.
  274. Contact: <sip:16034531989@192.168.2.111:5060>.
  275. Content-Type: application/sdp.
  276. Content-Length: 315.
  277. .
  278. v=0.
  279. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  280. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  281. c=IN IP4 192.168.2.111.
  282. t=0 0.
  283. m=audio 34030 RTP/AVP 8 0 101.
  284. a=rtpmap:8 PCMA/8000.
  285. a=rtpmap:0 PCMU/8000.
  286. a=rtpmap:101 telephone-event/8000.
  287. a=fmtp:101 0-16.
  288. a=silenceSupp:off - - - -.
  289. a=ptime:20.
  290. a=sendrecv.
  291.  
  292.  
  293. U 192.168.2.102:5060 -> 192.168.2.11:5060
  294. SIP/2.0 200 OK.
  295. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  296. Record-Route: <sip:opensips1.test.com;lr>.
  297. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  298. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  299. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  300. CSeq: 2 INVITE.
  301. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  302. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  303. Supported: replaces, timer.
  304. Contact: <sip:16034531989@192.168.2.111:5060>.
  305. Content-Type: application/sdp.
  306. Content-Length: 315.
  307. .
  308. v=0.
  309. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  310. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  311. c=IN IP4 192.168.2.111.
  312. t=0 0.
  313. m=audio 34030 RTP/AVP 8 0 101.
  314. a=rtpmap:8 PCMA/8000.
  315. a=rtpmap:0 PCMU/8000.
  316. a=rtpmap:101 telephone-event/8000.
  317. a=fmtp:101 0-16.
  318. a=silenceSupp:off - - - -.
  319. a=ptime:20.
  320. a=sendrecv.
  321.  
  322.  
  323. U 192.168.2.111:5060 -> 192.168.2.102:5060
  324. SIP/2.0 200 OK.
  325. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  326. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  327. Record-Route: <sip:opensips1.test.com;lr>.
  328. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  329. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  330. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  331. CSeq: 2 INVITE.
  332. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  333. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  334. Supported: replaces, timer.
  335. Contact: <sip:16034531989@192.168.2.111:5060>.
  336. Content-Type: application/sdp.
  337. Content-Length: 315.
  338. .
  339. v=0.
  340. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  341. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  342. c=IN IP4 192.168.2.111.
  343. t=0 0.
  344. m=audio 34030 RTP/AVP 8 0 101.
  345. a=rtpmap:8 PCMA/8000.
  346. a=rtpmap:0 PCMU/8000.
  347. a=rtpmap:101 telephone-event/8000.
  348. a=fmtp:101 0-16.
  349. a=silenceSupp:off - - - -.
  350. a=ptime:20.
  351. a=sendrecv.
  352.  
  353.  
  354. U 192.168.2.102:5060 -> 192.168.2.11:5060
  355. SIP/2.0 200 OK.
  356. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  357. Record-Route: <sip:opensips1.test.com;lr>.
  358. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  359. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  360. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  361. CSeq: 2 INVITE.
  362. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  363. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  364. Supported: replaces, timer.
  365. Contact: <sip:16034531989@192.168.2.111:5060>.
  366. Content-Type: application/sdp.
  367. Content-Length: 315.
  368. .
  369. v=0.
  370. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  371. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  372. c=IN IP4 192.168.2.111.
  373. t=0 0.
  374. m=audio 34030 RTP/AVP 8 0 101.
  375. a=rtpmap:8 PCMA/8000.
  376. a=rtpmap:0 PCMU/8000.
  377. a=rtpmap:101 telephone-event/8000.
  378. a=fmtp:101 0-16.
  379. a=silenceSupp:off - - - -.
  380. a=ptime:20.
  381. a=sendrecv.
  382.  
  383.  
  384. U 192.168.2.111:5060 -> 192.168.2.102:5060
  385. SIP/2.0 200 OK.
  386. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  387. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  388. Record-Route: <sip:opensips1.test.com;lr>.
  389. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  390. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  391. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  392. CSeq: 2 INVITE.
  393. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  394. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  395. Supported: replaces, timer.
  396. Contact: <sip:16034531989@192.168.2.111:5060>.
  397. Content-Type: application/sdp.
  398. Content-Length: 315.
  399. .
  400. v=0.
  401. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  402. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  403. c=IN IP4 192.168.2.111.
  404. t=0 0.
  405. m=audio 34030 RTP/AVP 8 0 101.
  406. a=rtpmap:8 PCMA/8000.
  407. a=rtpmap:0 PCMU/8000.
  408. a=rtpmap:101 telephone-event/8000.
  409. a=fmtp:101 0-16.
  410. a=silenceSupp:off - - - -.
  411. a=ptime:20.
  412. a=sendrecv.
  413.  
  414.  
  415. U 192.168.2.102:5060 -> 192.168.2.11:5060
  416. SIP/2.0 200 OK.
  417. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  418. Record-Route: <sip:opensips1.test.com;lr>.
  419. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  420. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  421. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  422. CSeq: 2 INVITE.
  423. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  424. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  425. Supported: replaces, timer.
  426. Contact: <sip:16034531989@192.168.2.111:5060>.
  427. Content-Type: application/sdp.
  428. Content-Length: 315.
  429. .
  430. v=0.
  431. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  432. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  433. c=IN IP4 192.168.2.111.
  434. t=0 0.
  435. m=audio 34030 RTP/AVP 8 0 101.
  436. a=rtpmap:8 PCMA/8000.
  437. a=rtpmap:0 PCMU/8000.
  438. a=rtpmap:101 telephone-event/8000.
  439. a=fmtp:101 0-16.
  440. a=silenceSupp:off - - - -.
  441. a=ptime:20.
  442. a=sendrecv.
  443.  
  444.  
  445. U 192.168.2.111:5060 -> 192.168.2.102:5060
  446. SIP/2.0 200 OK.
  447. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  448. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  449. Record-Route: <sip:opensips1.test.com;lr>.
  450. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  451. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  452. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  453. CSeq: 2 INVITE.
  454. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  455. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  456. Supported: replaces, timer.
  457. Contact: <sip:16034531989@192.168.2.111:5060>.
  458. Content-Type: application/sdp.
  459. Content-Length: 315.
  460. .
  461. v=0.
  462. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  463. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  464. c=IN IP4 192.168.2.111.
  465. t=0 0.
  466. m=audio 34030 RTP/AVP 8 0 101.
  467. a=rtpmap:8 PCMA/8000.
  468. a=rtpmap:0 PCMU/8000.
  469. a=rtpmap:101 telephone-event/8000.
  470. a=fmtp:101 0-16.
  471. a=silenceSupp:off - - - -.
  472. a=ptime:20.
  473. a=sendrecv.
  474.  
  475.  
  476. U 192.168.2.102:5060 -> 192.168.2.11:5060
  477. SIP/2.0 200 OK.
  478. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  479. Record-Route: <sip:opensips1.test.com;lr>.
  480. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  481. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  482. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  483. CSeq: 2 INVITE.
  484. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  485. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  486. Supported: replaces, timer.
  487. Contact: <sip:16034531989@192.168.2.111:5060>.
  488. Content-Type: application/sdp.
  489. Content-Length: 315.
  490. .
  491. v=0.
  492. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  493. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  494. c=IN IP4 192.168.2.111.
  495. t=0 0.
  496. m=audio 34030 RTP/AVP 8 0 101.
  497. a=rtpmap:8 PCMA/8000.
  498. a=rtpmap:0 PCMU/8000.
  499. a=rtpmap:101 telephone-event/8000.
  500. a=fmtp:101 0-16.
  501. a=silenceSupp:off - - - -.
  502. a=ptime:20.
  503. a=sendrecv.
  504.  
  505.  
  506. U 192.168.2.111:5060 -> 192.168.2.102:5060
  507. SIP/2.0 200 OK.
  508. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  509. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  510. Record-Route: <sip:opensips1.test.com;lr>.
  511. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  512. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  513. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  514. CSeq: 2 INVITE.
  515. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  516. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  517. Supported: replaces, timer.
  518. Contact: <sip:16034531989@192.168.2.111:5060>.
  519. Content-Type: application/sdp.
  520. Content-Length: 315.
  521. .
  522. v=0.
  523. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  524. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  525. c=IN IP4 192.168.2.111.
  526. t=0 0.
  527. m=audio 34030 RTP/AVP 8 0 101.
  528. a=rtpmap:8 PCMA/8000.
  529. a=rtpmap:0 PCMU/8000.
  530. a=rtpmap:101 telephone-event/8000.
  531. a=fmtp:101 0-16.
  532. a=silenceSupp:off - - - -.
  533. a=ptime:20.
  534. a=sendrecv.
  535.  
  536.  
  537. U 192.168.2.111:5060 -> 192.168.2.102:5060
  538. SIP/2.0 200 OK.
  539. Via: SIP/2.0/UDP opensips1.test.com;branch=z9hG4bKabcb.6fb8bb54.0;received=192.168.2.102.
  540. Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKa319fbb6291B866.
  541. Record-Route: <sip:opensips1.test.com;lr>.
  542. From: "Mike Peer" <sip:1001@opensips1.test.com>;tag=5560E120-59CFB995.
  543. To: <sip:16034531989@opensips1.test.com;user=phone>;tag=as7543e6d3.
  544. Call-ID: ccd49f54-6d2cf61a-278af3af@192.168.2.11.
  545. CSeq: 2 INVITE.
  546. Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
  547. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  548. Supported: replaces, timer.
  549. Contact: <sip:16034531989@192.168.2.111:5060>.
  550. Content-Type: application/sdp.
  551. Content-Length: 315.
  552. .
  553. v=0.
  554. o=root 1755367377 1755367377 IN IP4 192.168.2.111.
  555. s=Asterisk PBX UNKNOWN__and_probably_unsupported.
  556. c=IN IP4 192.168.2.111.
  557. t=0 0.
  558. m=audio 34030 RTP/AVP 8 0 101.
  559. a=rtpmap:8 PCMA/8000.
  560. a=rtpmap:0 PCMU/8000.
  561. a=rtpmap:101 telephone-event/8000.
  562. a=fmtp:101 0-16.
  563. a=silenceSupp:off - - - -.
  564. a=ptime:20.
  565. a=sendrecv.
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