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  1. <------------->
  2. --- (10 headers 0 lines) ---
  3. Really destroying SIP dialog '25dc84c458bc9eb334aa0f8f39a4e6f7@192.168.190.8:5060' Method: INVITE
  4. mars*CLI>
  5. == Using SIP RTP CoS mark 5
  6. -- Executing [002347081381170@office:1] Dial("SIP/2000-00000012", "SIP/terrasip-out/002347081381170") in new stack
  7. == Using SIP RTP CoS mark 5
  8. Audio is at 5060
  9. Adding codec 0x8 (alaw) to SDP
  10. Adding non-codec 0x1 (telephone-event) to SDP
  11. Reliably Transmitting (NAT) to 193.47.84.4:5060:
  12. INVITE sip:002347081381170@terrasip.net SIP/2.0
  13. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK1345c645;rport
  14. Max-Forwards: 70
  15. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  16. To: <sip:002347081381170@terrasip.net>
  17. Contact: <sip:21100053746@192.168.190.8:5060>
  18. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  19. CSeq: 102 INVITE
  20. User-Agent: Asterisk PBX 1.8.2.2
  21. Date: Tue, 01 Mar 2011 08:51:32 GMT
  22. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  23. Supported: replaces, timer
  24. Content-Type: application/sdp
  25. Content-Length: 236
  26.  
  27. v=0
  28. o=root 878504359 878504359 IN IP4 192.168.190.8
  29. s=Asterisk PBX 1.8.2.2
  30. c=IN IP4 192.168.190.8
  31. t=0 0
  32. m=audio 13484 RTP/AVP 8 101
  33. a=rtpmap:8 PCMA/8000
  34. a=rtpmap:101 telephone-event/8000
  35. a=fmtp:101 0-16
  36. a=ptime:20
  37. a=sendrecv
  38.  
  39. ---
  40. -- Called terrasip-out/002347081381170
  41.  
  42. <--- SIP read from UDP:193.47.84.4:5060 --->
  43. SIP/2.0 407 Proxy Authentication Required
  44. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK1345c645;rport=5060;received=82.128.127.2
  45. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  46. To: <sip:002347081381170@terrasip.net>;tag=598d351d4bc9e6e8e2c538995c9c64c8.22e5
  47. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  48. CSeq: 102 INVITE
  49. Proxy-Authenticate: Digest realm="terrasip.net", nonce="4d6cb3b2000060253b602cb9b144f3bcd6ba9af91ef6d57a"
  50. Server: TerraSip Advanced Router 1.0.16
  51. Content-Length: 0
  52.  
  53. <------------->
  54. --- (9 headers 0 lines) ---
  55. Transmitting (NAT) to 193.47.84.4:5060:
  56. ACK sip:002347081381170@terrasip.net SIP/2.0
  57. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK1345c645;rport
  58. Max-Forwards: 70
  59. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  60. To: <sip:002347081381170@terrasip.net>;tag=598d351d4bc9e6e8e2c538995c9c64c8.22e5
  61. Contact: <sip:21100053746@192.168.190.8:5060>
  62. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  63. CSeq: 102 ACK
  64. User-Agent: Asterisk PBX 1.8.2.2
  65. Content-Length: 0
  66.  
  67.  
  68. ---
  69. Audio is at 5060
  70. Adding codec 0x8 (alaw) to SDP
  71. Adding non-codec 0x1 (telephone-event) to SDP
  72. Reliably Transmitting (NAT) to 193.47.84.4:5060:
  73. INVITE sip:002347081381170@terrasip.net SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK7c717ded;rport
  75. Max-Forwards: 70
  76. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  77. To: <sip:002347081381170@terrasip.net>
  78. Contact: <sip:21100053746@192.168.190.8:5060>
  79. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  80. CSeq: 103 INVITE
  81. User-Agent: Asterisk PBX 1.8.2.2
  82. Proxy-Authorization: Digest username="21100053746", realm="terrasip.net", algorithm=MD5, uri="sip:002347081381170@terrasip.net", nonce="4d6cb3b2000060253b602cb9b144f3bcd6ba9af91ef6d57a", response="e2ba7830f81c0bbefb63fe469922eb31"
  83. Date: Tue, 01 Mar 2011 08:51:32 GMT
  84. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  85. Supported: replaces, timer
  86. Content-Type: application/sdp
  87. Content-Length: 236
  88.  
  89. v=0
  90. o=root 878504359 878504360 IN IP4 192.168.190.8
  91. s=Asterisk PBX 1.8.2.2
  92. c=IN IP4 192.168.190.8
  93. t=0 0
  94. m=audio 13484 RTP/AVP 8 101
  95. a=rtpmap:8 PCMA/8000
  96. a=rtpmap:101 telephone-event/8000
  97. a=fmtp:101 0-16
  98. a=ptime:20
  99. a=sendrecv
  100.  
  101. ---
  102.  
  103. <--- SIP read from UDP:193.47.84.4:5060 --->
  104. SIP/2.0 100 Giving a try
  105. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK7c717ded;rport=5060;received=82.128.127.2
  106. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  107. To: <sip:002347081381170@terrasip.net>
  108. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  109. CSeq: 103 INVITE
  110. Server: TerraSip Advanced Router 1.0.16
  111. Content-Length: 0
  112.  
  113. <------------->
  114. --- (8 headers 0 lines) ---
  115.  
  116. <--- SIP read from UDP:193.47.84.4:5060 --->
  117. SIP/2.0 183 Session Progress
  118. Via: SIP/2.0/UDP 192.168.190.8:5060;received=82.128.127.2;branch=z9hG4bK7c717ded;rport=5060
  119. Record-Route: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  120. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  121. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  122. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  123. CSeq: 103 INVITE
  124. User-Agent: gigasip-ffm56
  125. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  126. Supported: replaces
  127. Contact: <sip:002347081381170@195.20.217.56:5060;nat=yes>
  128. Content-Type: application/sdp
  129. Content-Length: 256
  130.  
  131. v=0
  132. o=root 24228 24228 IN IP4 193.47.84.4
  133. s=session
  134. c=IN IP4 193.47.84.4
  135. t=0 0
  136. m=audio 37866 RTP/AVP 8 101
  137. a=rtpmap:8 PCMA/8000
  138. a=rtpmap:101 telephone-event/8000
  139. a=fmtp:101 0-16
  140. a=silenceSupp:off - - - -
  141. a=ptime:20
  142. a=sendrecv
  143. a=nortpproxy:yes
  144. <------------->
  145. --- (13 headers 13 lines) ---
  146. Found RTP audio format 8
  147. Found RTP audio format 101
  148. Found audio description format PCMA for ID 8
  149. Found audio description format telephone-event for ID 101
  150. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  151. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  152. Peer audio RTP is at port 193.47.84.4:37866
  153. -- SIP/terrasip-out-00000013 is making progress passing it to SIP/2000-00000012
  154.  
  155. <--- SIP read from UDP:193.47.84.4:5060 --->
  156. SIP/2.0 200 OK
  157. Via: SIP/2.0/UDP 192.168.190.8:5060;received=82.128.127.2;branch=z9hG4bK7c717ded;rport=5060
  158. Record-Route: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  159. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  160. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  161. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  162. CSeq: 103 INVITE
  163. User-Agent: gigasip-ffm56
  164. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  165. Supported: replaces
  166. Contact: <sip:002347081381170@195.20.217.56:5060;nat=yes>
  167. Content-Type: application/sdp
  168. Content-Length: 256
  169.  
  170. v=0
  171. o=root 24228 24229 IN IP4 193.47.84.4
  172. s=session
  173. c=IN IP4 193.47.84.4
  174. t=0 0
  175. m=audio 37866 RTP/AVP 8 101
  176. a=rtpmap:8 PCMA/8000
  177. a=rtpmap:101 telephone-event/8000
  178. a=fmtp:101 0-16
  179. a=silenceSupp:off - - - -
  180. a=ptime:20
  181. a=sendrecv
  182. a=nortpproxy:yes
  183. <------------->
  184. --- (13 headers 13 lines) ---
  185. Found RTP audio format 8
  186. Found RTP audio format 101
  187. Found audio description format PCMA for ID 8
  188. Found audio description format telephone-event for ID 101
  189. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  190. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  191. Peer audio RTP is at port 193.47.84.4:37866
  192. list_route: hop: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  193. set_destination: Parsing <sip:193.47.84.4;lr=on;ftag=as435733b7> for address/port to send to
  194. set_destination: set destination to 193.47.84.4:5060
  195. Transmitting (NAT) to 193.47.84.4:5060:
  196. ACK sip:002347081381170@195.20.217.56:5060;nat=yes SIP/2.0
  197. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK53bf3d92;rport
  198. Route: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  199. Max-Forwards: 70
  200. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  201. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  202. Contact: <sip:21100053746@192.168.190.8:5060>
  203. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  204. CSeq: 103 ACK
  205. User-Agent: Asterisk PBX 1.8.2.2
  206. Content-Length: 0
  207.  
  208.  
  209. ---
  210. -- SIP/terrasip-out-00000013 answered SIP/2000-00000012
  211. -- Remotely bridging SIP/2000-00000012 and SIP/terrasip-out-00000013
  212. set_destination: Parsing <sip:193.47.84.4;lr=on;ftag=as435733b7> for address/port to send to
  213. set_destination: set destination to 193.47.84.4:5060
  214. Audio is at 5060
  215. Adding codec 0x8 (alaw) to SDP
  216. Adding non-codec 0x1 (telephone-event) to SDP
  217. Reliably Transmitting (NAT) to 193.47.84.4:5060:
  218. INVITE sip:002347081381170@195.20.217.56:5060;nat=yes SIP/2.0
  219. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK1a050d59;rport
  220. Route: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  221. Max-Forwards: 70
  222. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  223. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  224. Contact: <sip:21100053746@192.168.190.8:5060>
  225. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  226. CSeq: 104 INVITE
  227. User-Agent: Asterisk PBX 1.8.2.2
  228. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  229. Supported: replaces, timer
  230. X-asterisk-Info: SIP re-invite (External RTP bridge)
  231. Content-Type: application/sdp
  232. Content-Length: 238
  233.  
  234. v=0
  235. o=root 878504359 878504361 IN IP4 192.168.124.36
  236. s=Asterisk PBX 1.8.2.2
  237. c=IN IP4 192.168.124.36
  238. t=0 0
  239. m=audio 10600 RTP/AVP 8 101
  240. a=rtpmap:8 PCMA/8000
  241. a=rtpmap:101 telephone-event/8000
  242. a=fmtp:101 0-16
  243. a=ptime:20
  244. a=sendrecv
  245.  
  246. ---
  247.  
  248. <--- SIP read from UDP:193.47.84.4:5060 --->
  249. SIP/2.0 100 Giving a try
  250. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK1a050d59;rport=5060;received=82.128.127.2
  251. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  252. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  253. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  254. CSeq: 104 INVITE
  255. Server: TerraSip Advanced Router 1.0.16
  256. Content-Length: 0
  257.  
  258. <------------->
  259. --- (8 headers 0 lines) ---
  260.  
  261. <--- SIP read from UDP:193.47.84.4:5060 --->
  262. SIP/2.0 200 OK
  263. Via: SIP/2.0/UDP 192.168.190.8:5060;received=82.128.127.2;branch=z9hG4bK1a050d59;rport=5060
  264. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  265. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  266. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  267. CSeq: 104 INVITE
  268. User-Agent: gigasip-ffm56
  269. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  270. Supported: replaces
  271. Contact: <sip:002347081381170@195.20.217.56:5060;nat=yes>
  272. Content-Type: application/sdp
  273. Content-Length: 256
  274.  
  275. v=0
  276. o=root 24228 24230 IN IP4 193.47.84.4
  277. s=session
  278. c=IN IP4 193.47.84.4
  279. t=0 0
  280. m=audio 37866 RTP/AVP 8 101
  281. a=rtpmap:8 PCMA/8000
  282. a=rtpmap:101 telephone-event/8000
  283. a=fmtp:101 0-16
  284. a=silenceSupp:off - - - -
  285. a=ptime:20
  286. a=sendrecv
  287. a=nortpproxy:yes
  288. <------------->
  289. --- (12 headers 13 lines) ---
  290. Found RTP audio format 8
  291. Found RTP audio format 101
  292. Found audio description format PCMA for ID 8
  293. Found audio description format telephone-event for ID 101
  294. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  295. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  296. Peer audio RTP is at port 193.47.84.4:37866
  297. set_destination: Parsing <sip:193.47.84.4;lr=on;ftag=as435733b7> for address/port to send to
  298. set_destination: set destination to 193.47.84.4:5060
  299. Transmitting (NAT) to 193.47.84.4:5060:
  300. ACK sip:002347081381170@195.20.217.56:5060;nat=yes SIP/2.0
  301. Via: SIP/2.0/UDP 192.168.190.8:5060;branch=z9hG4bK3b5c19fe;rport
  302. Route: <sip:193.47.84.4;lr=on;ftag=as435733b7>
  303. Max-Forwards: 70
  304. From: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  305. To: <sip:002347081381170@terrasip.net>;tag=as023d622b
  306. Contact: <sip:21100053746@192.168.190.8:5060>
  307. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  308. CSeq: 104 ACK
  309. User-Agent: Asterisk PBX 1.8.2.2
  310. Content-Length: 0
  311.  
  312.  
  313. ---
  314.  
  315. <--- SIP read from UDP:193.47.84.4:5060 --->
  316. BYE sip:21100053746@82.128.127.2:5060;nat=yes SIP/2.0
  317. Via: SIP/2.0/UDP 193.47.84.4;branch=z9hG4bKa44d.a13875a3.0
  318. Via: SIP/2.0/UDP 195.20.217.56:5060;received=195.20.217.56;branch=z9hG4bK15c949e3;rport=5060
  319. From: <sip:002347081381170@terrasip.net>;tag=as023d622b
  320. To: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  321. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  322. CSeq: 102 BYE
  323. User-Agent: gigasip-ffm56
  324. Max-Forwards: 69
  325. X-Asterisk-HangupCause: Normal Clearing
  326. X-Asterisk-HangupCauseCode: 16
  327. Content-Length: 0
  328.  
  329. <------------->
  330. --- (12 headers 0 lines) ---
  331. Sending to 193.47.84.4:5060 (NAT)
  332. Scheduling destruction of SIP dialog '39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060' in 32000 ms (Method: BYE)
  333.  
  334. <--- Transmitting (NAT) to 193.47.84.4:5060 --->
  335. SIP/2.0 200 OK
  336. Via: SIP/2.0/UDP 193.47.84.4;branch=z9hG4bKa44d.a13875a3.0;received=193.47.84.4;rport=5060
  337. Via: SIP/2.0/UDP 195.20.217.56:5060;received=195.20.217.56;branch=z9hG4bK15c949e3;rport=5060
  338. From: <sip:002347081381170@terrasip.net>;tag=as023d622b
  339. To: "Scott Home" <sip:21100053746@192.168.190.8>;tag=as435733b7
  340. Call-ID: 39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060
  341. CSeq: 102 BYE
  342. Server: Asterisk PBX 1.8.2.2
  343. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  344. Supported: replaces, timer
  345. Content-Length: 0
  346.  
  347.  
  348. <------------>
  349. == Spawn extension (office, 002347081381170, 1) exited non-zero on 'SIP/2000-00000012'
  350. Really destroying SIP dialog '39002d8e41bdfd696bb734ea2ad2ff9a@192.168.190.8:5060' Method: BYE
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