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  1. <--- SIP read from UDP:95.26.225.1:5060 --->
  2. INVITE sip:79250287897@sip.mydomain.lol SIP/2.0
  3. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK38f63e8d;rport
  4. Max-Forwards: 70
  5. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  6. To: <sip:79250287897@sip.mydomain.lol>
  7. Contact: <sip:user33@95.26.225.1:5060>
  8. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  9. CSeq: 102 INVITE
  10. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  11. Date: Mon, 30 Apr 2012 11:54:40 GMT
  12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  13. Supported: replaces, timer
  14. Content-Type: application/sdp
  15. Content-Length: 269
  16.  
  17. v=0
  18. o=root 1984994344 1984994344 IN IP4 95.26.225.1
  19. s=Asterisk PBX 1.8.10.1~dfsg-1+b1
  20. c=IN IP4 95.26.225.1
  21. t=0 0
  22. m=audio 10080 RTP/AVP 0 8 101
  23. a=rtpmap:0 PCMU/8000
  24. a=rtpmap:8 PCMA/8000
  25. a=rtpmap:101 telephone-event/8000
  26. a=fmtp:101 0-16
  27. a=ptime:20
  28. a=sendrecv
  29. <------------->
  30. --- (14 headers 12 lines) ---
  31. Sending to 95.26.225.1:5060 (NAT)
  32. Using INVITE request as basis request - 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  33. Found peer 'root' for 'user33' from 95.26.225.1:5060
  34.  
  35. <--- Reliably Transmitting (NAT) to 95.26.225.1:5060 --->
  36. SIP/2.0 401 Unauthorized
  37. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK38f63e8d;received=95.26.225.1;rport=5060
  38. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  39. To: <sip:79250287897@sip.mydomain.lol>;tag=as0e3b4ddd
  40. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  41. CSeq: 102 INVITE
  42. Server: SipPhone
  43. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  44. Supported: replaces, timer
  45. WWW-Authenticate: Digest algorithm=MD5, realm="ster", nonce="2493c04d"
  46. Content-Length: 0
  47.  
  48.  
  49. <------------>
  50. Scheduling destruction of SIP dialog '12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060' in 6400 ms (Method: INVITE)
  51.  
  52. <--- SIP read from UDP:95.26.225.1:5060 --->
  53. ACK sip:79250287897@sip.mydomain.lol SIP/2.0
  54. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK38f63e8d;rport
  55. Max-Forwards: 70
  56. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  57. To: <sip:79250287897@sip.mydomain.lol>;tag=as0e3b4ddd
  58. Contact: <sip:user33@95.26.225.1:5060>
  59. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  60. CSeq: 102 ACK
  61. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  62. Content-Length: 0
  63.  
  64. <------------->
  65. --- (10 headers 0 lines) ---
  66.  
  67. <--- SIP read from UDP:95.26.225.1:5060 --->
  68. INVITE sip:79250287897@sip.mydomain.lol SIP/2.0
  69. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK7dae4ace;rport
  70. Max-Forwards: 70
  71. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  72. To: <sip:79250287897@sip.mydomain.lol>
  73. Contact: <sip:user33@95.26.225.1:5060>
  74. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  75. CSeq: 103 INVITE
  76. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  77. Authorization: Digest username="root", realm="ster", algorithm=MD5, uri="sip:79250287897@sip.mydomain.lol", nonce="2493c04d", response="154a7c7daf2ef8a82fe7e0c6b8ea2a86"
  78. Date: Mon, 30 Apr 2012 11:54:40 GMT
  79. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  80. Supported: replaces, timer
  81. Content-Type: application/sdp
  82. Content-Length: 269
  83.  
  84. v=0
  85. o=root 1984994344 1984994345 IN IP4 95.26.225.1
  86. s=Asterisk PBX 1.8.10.1~dfsg-1+b1
  87. c=IN IP4 95.26.225.1
  88. t=0 0
  89. m=audio 10080 RTP/AVP 0 8 101
  90. a=rtpmap:0 PCMU/8000
  91. a=rtpmap:8 PCMA/8000
  92. a=rtpmap:101 telephone-event/8000
  93. a=fmtp:101 0-16
  94. a=ptime:20
  95. a=sendrecv
  96. <------------->
  97. --- (15 headers 12 lines) ---
  98. Sending to 95.26.225.1:5060 (NAT)
  99. Using INVITE request as basis request - 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  100. Found peer 'root' for 'user33' from 95.26.225.1:5060
  101. Found RTP audio format 0
  102. Found RTP audio format 8
  103. Found RTP audio format 101
  104. Found audio description format PCMU for ID 0
  105. Found audio description format PCMA for ID 8
  106. Found audio description format telephone-event for ID 101
  107. Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  108. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  109. Peer audio RTP is at port 95.26.225.1:10080
  110. Looking for 79250287897 in sip (domain sip.mydomain.lol)
  111. list_route: hop: <sip:user33@95.26.225.1:5060>
  112.  
  113. <--- Transmitting (NAT) to 95.26.225.1:5060 --->
  114. SIP/2.0 100 Trying
  115. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK7dae4ace;received=95.26.225.1;rport=5060
  116. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  117. To: <sip:79250287897@sip.mydomain.lol>
  118. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  119. CSeq: 103 INVITE
  120. Server: SipPhone
  121. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  122. Supported: replaces, timer
  123. Contact: <sip:79250287897@66.66.66.66:5060>
  124. Content-Length: 0
  125.  
  126.  
  127. <------------>
  128.  
  129. <--- Transmitting (NAT) to 95.26.225.1:5060 --->
  130. SIP/2.0 180 Ringing
  131. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK7dae4ace;received=95.26.225.1;rport=5060
  132. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  133. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  134. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  135. CSeq: 103 INVITE
  136. Server: SipPhone
  137. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  138. Supported: replaces, timer
  139. Contact: <sip:79250287897@66.66.66.66:5060>
  140. Content-Length: 0
  141.  
  142.  
  143. <------------>
  144. Audio is at 19118
  145. Adding codec 100003 (ulaw) to SDP
  146. Adding codec 100004 (alaw) to SDP
  147. Adding non-codec 0x1 (telephone-event) to SDP
  148.  
  149. <--- Reliably Transmitting (NAT) to 95.26.225.1:5060 --->
  150. SIP/2.0 200 OK
  151. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK7dae4ace;received=95.26.225.1;rport=5060
  152. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  153. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  154. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  155. CSeq: 103 INVITE
  156. Server: SipPhone
  157. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  158. Supported: replaces, timer
  159. Contact: <sip:79250287897@66.66.66.66:5060>
  160. Content-Type: application/sdp
  161. Content-Length: 263
  162.  
  163. v=0
  164. o=root 859731047 859731047 IN IP4 66.66.66.66
  165. s=Asterisk PBX 10.4.0-rc2
  166. c=IN IP4 66.66.66.66
  167. t=0 0
  168. m=audio 19118 RTP/AVP 0 8 101
  169. a=rtpmap:0 PCMU/8000
  170. a=rtpmap:8 PCMA/8000
  171. a=rtpmap:101 telephone-event/8000
  172. a=fmtp:101 0-16
  173. a=ptime:20
  174. a=sendrecv
  175.  
  176. <------------>
  177.  
  178. <--- SIP read from UDP:95.26.225.1:5060 --->
  179. ACK sip:79250287897@66.66.66.66:5060 SIP/2.0
  180. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK68d0175c;rport
  181. Max-Forwards: 70
  182. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  183. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  184. Contact: <sip:user33@95.26.225.1:5060>
  185. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  186. CSeq: 103 ACK
  187. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  188. Content-Length: 0
  189.  
  190. <------------->
  191. --- (10 headers 0 lines) ---
  192.  
  193. <--- SIP read from UDP:95.26.225.1:5060 --->
  194. INVITE sip:79250287897@66.66.66.66:5060 SIP/2.0
  195. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK296fe192;rport
  196. Max-Forwards: 70
  197. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  198. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  199. Contact: <sip:user33@95.26.225.1:5060>
  200. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  201. CSeq: 104 INVITE
  202. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  203. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  204. Supported: replaces, timer
  205. Content-Type: application/sdp
  206. Content-Length: 269
  207.  
  208. v=0
  209. o=root 1984994344 1984994346 IN IP4 95.26.225.1
  210. s=Asterisk PBX 1.8.10.1~dfsg-1+b1
  211. c=IN IP4 95.26.225.1
  212. t=0 0
  213. m=audio 10600 RTP/AVP 0 8 101
  214. a=rtpmap:0 PCMU/8000
  215. a=rtpmap:8 PCMA/8000
  216. a=rtpmap:101 telephone-event/8000
  217. a=fmtp:101 0-16
  218. a=ptime:20
  219. a=sendrecv
  220. <------------->
  221. --- (13 headers 12 lines) ---
  222. Sending to 95.26.225.1:5060 (NAT)
  223. Found RTP audio format 0
  224. Found RTP audio format 8
  225. Found RTP audio format 101
  226. Found audio description format PCMU for ID 0
  227. Found audio description format PCMA for ID 8
  228. Found audio description format telephone-event for ID 101
  229. Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  230. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  231. Peer audio RTP is at port 95.26.225.1:10600
  232.  
  233. <--- Transmitting (NAT) to 95.26.225.1:5060 --->
  234. SIP/2.0 100 Trying
  235. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK296fe192;received=95.26.225.1;rport=5060
  236. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  237. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  238. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  239. CSeq: 104 INVITE
  240. Server: SipPhone
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  242. Supported: replaces, timer
  243. Contact: <sip:79250287897@66.66.66.66:5060>
  244. Content-Length: 0
  245.  
  246.  
  247. <------------>
  248. Audio is at 19118
  249. Adding codec 100003 (ulaw) to SDP
  250. Adding codec 100004 (alaw) to SDP
  251. Adding non-codec 0x1 (telephone-event) to SDP
  252.  
  253. <--- Reliably Transmitting (NAT) to 95.26.225.1:5060 --->
  254. SIP/2.0 200 OK
  255. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK296fe192;received=95.26.225.1;rport=5060
  256. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  257. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  258. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  259. CSeq: 104 INVITE
  260. Server: SipPhone
  261. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  262. Supported: replaces, timer
  263. Contact: <sip:79250287897@66.66.66.66:5060>
  264. Content-Type: application/sdp
  265. Content-Length: 263
  266.  
  267. v=0
  268. o=root 859731047 859731048 IN IP4 66.66.66.66
  269. s=Asterisk PBX 10.4.0-rc2
  270. c=IN IP4 66.66.66.66
  271. t=0 0
  272. m=audio 19118 RTP/AVP 0 8 101
  273. a=rtpmap:0 PCMU/8000
  274. a=rtpmap:8 PCMA/8000
  275. a=rtpmap:101 telephone-event/8000
  276. a=fmtp:101 0-16
  277. a=ptime:20
  278. a=sendrecv
  279.  
  280. <------------>
  281.  
  282. <--- SIP read from UDP:95.26.225.1:5060 --->
  283. ACK sip:79250287897@66.66.66.66:5060 SIP/2.0
  284. Via: SIP/2.0/UDP 95.26.225.1:5060;branch=z9hG4bK71966f04;rport
  285. Max-Forwards: 70
  286. From: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  287. To: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  288. Contact: <sip:user33@95.26.225.1:5060>
  289. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  290. CSeq: 104 ACK
  291. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1+b1
  292. Content-Length: 0
  293.  
  294. <------------->
  295. --- (10 headers 0 lines) ---
  296. Scheduling destruction of SIP dialog '12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060' in 6400 ms (Method: ACK)
  297. set_destination: Parsing <sip:user33@95.26.225.1:5060> for address/port to send to
  298. set_destination: set destination to 95.26.225.1:5060
  299. Reliably Transmitting (NAT) to 95.26.225.1:5060:
  300. BYE sip:user33@95.26.225.1:5060 SIP/2.0
  301. Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK021541c3;rport
  302. Max-Forwards: 70
  303. From: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  304. To: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  305. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  306. CSeq: 102 BYE
  307. User-Agent: SipPhone
  308. Proxy-Authorization: Digest username="root", realm="ster", algorithm=MD5, uri="sip:sip.mydomain.lol", nonce="", response="1eaf83c6194a3c55299d347b99fe4fa9"
  309. X-Asterisk-HangupCause: Normal Clearing
  310. X-Asterisk-HangupCauseCode: 16
  311. Content-Length: 0
  312.  
  313.  
  314. ---
  315.  
  316. <--- SIP read from UDP:95.26.225.1:5060 --->
  317. SIP/2.0 200 OK
  318. Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK021541c3;received=66.66.66.66;rport=5060
  319. From: <sip:79250287897@sip.mydomain.lol>;tag=as50e0867a
  320. To: "user33" <sip:user33@95.26.225.1:5060>;tag=as3b09a95b
  321. Call-ID: 12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060
  322. CSeq: 102 BYE
  323. Server: Asterisk PBX 1.8.10.1~dfsg-1+b1
  324. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  325. Supported: replaces, timer
  326. Content-Length: 0
  327.  
  328. <------------->
  329. --- (10 headers 0 lines) ---
  330. SIP Response message for INCOMING dialog BYE arrived
  331. Really destroying SIP dialog '12c0f61f2a5d3fd50d19b0e75f02e735@192.168.0.25:5060' Method: ACK
  332. Really destroying SIP dialog '2df057d8337d65912fc337ea1e6a5b0c@127.0.1.1' Method: REGISTER
  333. ster*CLI>
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