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  1. home*CLI>
  2. sip set debug on
  3. SIP Debugging enabled
  4. home*CLI>
  5. <--- SIP read from UDP:198.65.166.131:5060 --->
  6. INVITE sip:s@98.226.XXX.XXX SIP/2.0
  7. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  8. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  9. Record-Route: <sip:74.125.64.80:24000;lr=on>
  10. Max-Forwards: 17
  11. To: <sip:747317XXXX@74.125.64.80:24000>
  12. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.e6881ab6.0
  13. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.d6881ab6.0
  14. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK7922.93dafb66.0
  15. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK2102910839
  16. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  17. Call-ID: 277698478@10.218.63.140
  18. CSeq: 68661 INVITE
  19. User-Agent: YATE/3.0.0
  20. Contact: <sip:+18008675309@74.125.64.80:24000>
  21. Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
  22. Content-Type: application/sdp
  23. Content-Length: 183
  24. RemoteIP: 74.125.64.80
  25. X-GoogleVoice: true
  26. P-hint: local number (2)
  27. P-Contact: 4
  28. Subject: via Google Voice
  29.  
  30. v=0
  31. o=yate 1283835844 1283835844 IN IP4 74.125.64.87
  32. s=SIP Call
  33. c=IN IP4 74.125.64.87
  34. t=0 0
  35. m=audio 25796 RTP/AVP 0 101
  36. a=rtpmap:0 PCMU/8000
  37. a=rtpmap:101 telephone-event/8000
  38.  
  39. <------------->
  40. --- (23 headers 8 lines) ---
  41. Sending to 198.65.166.131 : 5060 (NAT)
  42. Using INVITE request as basis request - 277698478@10.218.63.140
  43. Found peer 'proxy01.sipphone.com' for '+18008675309' from 198.65.166.131:5060
  44. Found RTP audio format 0
  45. Found RTP audio format 101
  46. Found audio description format PCMU for ID 0
  47. Found audio description format telephone-event for ID 101
  48. Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  49. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  50. Peer audio RTP is at port 74.125.64.87:25796
  51. Looking for s in default (domain 98.226.XXX.XXX)
  52. list_route: hop: <sip:198.65.166.131;lr;ftag=1422683903>
  53. list_route: hop: <sip:198.65.166.131;lr;ftag=1422683903>
  54. list_route: hop: <sip:74.125.64.80:24000;lr=on>
  55. home*CLI>
  56. <--- Transmitting (no NAT) to 198.65.166.131:5060 --->
  57. SIP/2.0 100 Trying
  58. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.e6881ab6.0;received=198.65.166.131
  59. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.d6881ab6.0
  60. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK7922.93dafb66.0
  61. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK2102910839
  62. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  63. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  64. Record-Route: <sip:74.125.64.80:24000;lr=on>
  65. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  66. To: <sip:747317XXXX@74.125.64.80:24000>
  67. Call-ID: 277698478@10.218.63.140
  68. CSeq: 68661 INVITE
  69. Server: Asterisk PBX 1.6.2.5-0ubuntu1
  70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  71. Supported: replaces, timer
  72. Contact: <sip:s@98.226.XXX.XXX>
  73. Content-Length: 0
  74.  
  75.  
  76. <------------>
  77. Audio is at 98.226.XXX.XXX port 16246
  78. Adding codec 0x4 (ulaw) to SDP
  79. Adding non-codec 0x1 (telephone-event) to SDP
  80.  
  81. <--- Reliably Transmitting (no NAT) to 198.65.166.131:5060 --->
  82. SIP/2.0 200 OK
  83. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.e6881ab6.0;received=198.65.166.131
  84. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.d6881ab6.0
  85. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK7922.93dafb66.0
  86. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK2102910839
  87. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  88. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  89. Record-Route: <sip:74.125.64.80:24000;lr=on>
  90. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  91. To: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  92. Call-ID: 277698478@10.218.63.140
  93. CSeq: 68661 INVITE
  94. Server: Asterisk PBX 1.6.2.5-0ubuntu1
  95. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  96. Supported: replaces, timer
  97. Contact: <sip:s@98.226.XXX.XXX>
  98. Content-Type: application/sdp
  99. Content-Length: 274
  100.  
  101. v=0
  102. o=root 132443849 132443849 IN IP4 98.226.XXX.XXX
  103. s=Asterisk PBX 1.6.2.5-0ubuntu1
  104. c=IN IP4 98.226.XXX.XXX
  105. t=0 0
  106. m=audio 16246 RTP/AVP 0 101
  107. a=rtpmap:0 PCMU/8000
  108. a=rtpmap:101 telephone-event/8000
  109. a=fmtp:101 0-16
  110. a=silenceSupp:off - - - -
  111. a=ptime:20
  112. a=sendrecv
  113.  
  114. <------------>
  115. Audio is at 192.168.2.250 port 18902
  116. Adding codec 0x4 (ulaw) to SDP
  117. Adding codec 0x8 (alaw) to SDP
  118. Adding non-codec 0x1 (telephone-event) to SDP
  119. Reliably Transmitting (no NAT) to 192.168.2.101:5060:
  120. INVITE sip:45@192.168.2.101:5060 SIP/2.0
  121. Via: SIP/2.0/UDP 192.168.2.250:5060;branch=z9hG4bK25ae10d6;rport
  122. Max-Forwards: 70
  123. From: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  124. To: <sip:45@192.168.2.101:5060>
  125. Contact: <sip:+18008675309@192.168.2.250>
  126. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  127. CSeq: 102 INVITE
  128. User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
  129. Date: Tue, 07 Sep 2010 05:04:07 GMT
  130. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  131. Supported: replaces, timer
  132. Content-Type: application/sdp
  133. Content-Length: 298
  134.  
  135. v=0
  136. o=root 1041293262 1041293262 IN IP4 192.168.2.250
  137. s=Asterisk PBX 1.6.2.5-0ubuntu1
  138. c=IN IP4 192.168.2.250
  139. t=0 0
  140. m=audio 18902 RTP/AVP 0 8 101
  141. a=rtpmap:0 PCMU/8000
  142. a=rtpmap:8 PCMA/8000
  143. a=rtpmap:101 telephone-event/8000
  144. a=fmtp:101 0-16
  145. a=silenceSupp:off - - - -
  146. a=ptime:20
  147. a=sendrecv
  148.  
  149. ---
  150. home*CLI>
  151. <--- SIP read from UDP:192.168.2.101:5060 --->
  152. SIP/2.0 100 Trying
  153. To: <sip:45@192.168.2.101:5060>
  154. From: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  155. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  156. CSeq: 102 INVITE
  157. Via: SIP/2.0/UDP 192.168.2.250:5060;branch=z9hG4bK25ae10d6
  158. Server: Linksys/PAP2T-3.1.15(LS)
  159. Content-Length: 0
  160.  
  161.  
  162. <------------->
  163. --- (8 headers 0 lines) ---
  164. home*CLI>
  165. <--- SIP read from UDP:192.168.2.101:5060 --->
  166. SIP/2.0 180 Ringing
  167. To: <sip:45@192.168.2.101:5060>;tag=c8296f50cdff8ed6i0
  168. From: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  169. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  170. CSeq: 102 INVITE
  171. Via: SIP/2.0/UDP 192.168.2.250:5060;branch=z9hG4bK25ae10d6
  172. Server: Linksys/PAP2T-3.1.15(LS)
  173. Content-Length: 0
  174.  
  175.  
  176. <------------->
  177. --- (8 headers 0 lines) ---
  178. home*CLI>
  179. <--- SIP read from UDP:198.65.166.131:5060 --->
  180. ACK sip:s@98.226.XXX.XXX:5060 SIP/2.0
  181. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  182. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  183. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.d6881ab6.2
  184. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK7922.d6881ab6.2
  185. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK7922.93dafb66.2
  186. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK1851441164
  187. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  188. To: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  189. Call-ID: 277698478@10.218.63.140
  190. CSeq: 68661 ACK
  191. Max-Forwards: 17
  192. Contact: <sip:+18008675309@10.218.63.140:7654>
  193. User-Agent: YATE/3.0.0
  194. Content-Length: 0
  195. P-hint: rr-enforced
  196. P-hint: rr-enforced
  197.  
  198.  
  199. <------------->
  200. --- (17 headers 0 lines) ---
  201. home*CLI>
  202. <--- SIP read from UDP:192.168.2.101:5060 --->
  203. SUBSCRIBE sip:192.168.2.250 SIP/2.0
  204. Via: SIP/2.0/UDP 192.168.2.101:5060;branch=z9hG4bK-fd33a245
  205. From: 45 <sip:45@192.168.2.250>;tag=b4e3a165894e7bfd
  206. To: 45 <sip:45@192.168.2.250>;tag=as71b279ff
  207. Call-ID: 98627b85-60c0685d@192.168.2.101
  208. CSeq: 14502 SUBSCRIBE
  209. Max-Forwards: 70
  210. Authorization: Digest username="45",realm="asterisk",nonce="1182db2e",uri="sip:192.168.2.250",algorithm=MD5,response="ee8c34d085808f4db4a2d5cc9f1cafc5"
  211. Contact: 45 <sip:45@192.168.2.101:5060>
  212. Accept: application/simple-message-summary
  213. Expires: 2147483647
  214. Event: message-summary
  215. User-Agent: Linksys/PAP2T-3.1.15(LS)
  216. Content-Length: 0
  217.  
  218.  
  219. <------------->
  220. --- (14 headers 0 lines) ---
  221. Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
  222.  
  223. <--- Transmitting (NAT) to 192.168.2.101:5060 --->
  224. SIP/2.0 481 Subscription does not exist
  225. Via: SIP/2.0/UDP 192.168.2.101:5060;branch=z9hG4bK-fd33a245;received=192.168.2.101
  226. From: 45 <sip:45@192.168.2.250>;tag=b4e3a165894e7bfd
  227. To: 45 <sip:45@192.168.2.250>;tag=as71b279ff
  228. Call-ID: 98627b85-60c0685d@192.168.2.101
  229. CSeq: 14502 SUBSCRIBE
  230. Server: Asterisk PBX 1.6.2.5-0ubuntu1
  231. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  232. Supported: replaces, timer
  233. Content-Length: 0
  234.  
  235.  
  236. <------------>
  237. Really destroying SIP dialog '98627b85-60c0685d@192.168.2.101' Method: SUBSCRIBE
  238. home*CLI>
  239. <--- SIP read from UDP:192.168.2.101:5060 --->
  240. SIP/2.0 200 OK
  241. To: <sip:45@192.168.2.101:5060>;tag=c8296f50cdff8ed6i0
  242. From: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  243. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  244. CSeq: 102 INVITE
  245. Via: SIP/2.0/UDP 192.168.2.250:5060;branch=z9hG4bK25ae10d6
  246. Contact: 45 <sip:45@192.168.2.101:5060>
  247. Server: Linksys/PAP2T-3.1.15(LS)
  248. Content-Length: 255
  249. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  250. Supported: x-sipura
  251. Content-Type: application/sdp
  252.  
  253. v=0
  254. o=- 885607 885607 IN IP4 192.168.2.101
  255. s=-
  256. c=IN IP4 192.168.2.101
  257. t=0 0
  258. m=audio 16436 RTP/AVP 0 100 101
  259. a=rtpmap:0 PCMU/8000
  260. a=rtpmap:100 NSE/8000
  261. a=fmtp:100 192-193
  262. a=rtpmap:101 telephone-event/8000
  263. a=fmtp:101 0-15
  264. a=ptime:30
  265. a=sendrecv
  266.  
  267. <------------->
  268. --- (12 headers 13 lines) ---
  269. Found RTP audio format 0
  270. Found RTP audio format 100
  271. Found RTP audio format 101
  272. Found audio description format PCMU for ID 0
  273. Found audio description format NSE for ID 100
  274. Found audio description format telephone-event for ID 101
  275. Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  276. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  277. Peer audio RTP is at port 192.168.2.101:16436
  278. list_route: hop: <sip:45@192.168.2.101:5060>
  279. set_destination: Parsing <sip:45@192.168.2.101:5060> for address/port to send to
  280. set_destination: set destination to 192.168.2.101, port 5060
  281. Transmitting (no NAT) to 192.168.2.101:5060:
  282. ACK sip:45@192.168.2.101:5060 SIP/2.0
  283. Via: SIP/2.0/UDP 192.168.2.250:5060;branch=z9hG4bK340f886f;rport
  284. Max-Forwards: 70
  285. From: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  286. To: <sip:45@192.168.2.101:5060>;tag=c8296f50cdff8ed6i0
  287. Contact: <sip:+18008675309@192.168.2.250>
  288. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  289. CSeq: 102 ACK
  290. User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
  291. Content-Length: 0
  292.  
  293.  
  294. ---
  295. home*CLI>
  296. <--- SIP read from UDP:192.168.2.101:5060 --->
  297. BYE sip:+18008675309@192.168.2.250 SIP/2.0
  298. Via: SIP/2.0/UDP 192.168.2.101:5060;branch=z9hG4bK-93637479
  299. From: <sip:45@192.168.2.101:5060>;tag=c8296f50cdff8ed6i0
  300. To: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  301. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  302. CSeq: 101 BYE
  303. Max-Forwards: 70
  304. User-Agent: Linksys/PAP2T-3.1.15(LS)
  305. Content-Length: 0
  306.  
  307.  
  308. <------------->
  309. --- (9 headers 0 lines) ---
  310. Sending to 192.168.2.101 : 5060 (no NAT)
  311. home*CLI>
  312. <--- Transmitting (no NAT) to 192.168.2.101:5060 --->
  313. SIP/2.0 200 OK
  314. Via: SIP/2.0/UDP 192.168.2.101:5060;branch=z9hG4bK-93637479;received=192.168.2.101
  315. From: <sip:45@192.168.2.101:5060>;tag=c8296f50cdff8ed6i0
  316. To: "+18008675309" <sip:+18008675309@192.168.2.250>;tag=as436bf732
  317. Call-ID: 24257976032b21900b3a862e5b2e186e@192.168.2.250
  318. CSeq: 101 BYE
  319. Server: Asterisk PBX 1.6.2.5-0ubuntu1
  320. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  321. Supported: replaces, timer
  322. Content-Length: 0
  323.  
  324.  
  325. <------------>
  326. Scheduling destruction of SIP dialog '277698478@10.218.63.140' in 7936 ms (Method: ACK)
  327. set_destination: Parsing <sip:198.65.166.131;lr;ftag=1422683903> for address/port to send to
  328. set_destination: set destination to 198.65.166.131, port 5060
  329. Reliably Transmitting (no NAT) to 198.65.166.131:5060:
  330. BYE sip:+18008675309@74.125.64.80:24000 SIP/2.0
  331. Via: SIP/2.0/UDP 98.226.XXX.XXX:5060;branch=z9hG4bK4bc0cb5c;rport
  332. Route: <sip:198.65.166.131;lr;ftag=1422683903>,<sip:198.65.166.131;lr;ftag=1422683903>,<sip:74.125.64.80:24000;lr=on>
  333. Max-Forwards: 70
  334. From: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  335. To: <sip:+18008675309@10.218.63.140>;tag=1422683903
  336. Call-ID: 277698478@10.218.63.140
  337. CSeq: 102 BYE
  338. User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
  339. X-Asterisk-HangupCause: Normal Clearing
  340. X-Asterisk-HangupCauseCode: 16
  341. Content-Length: 0
  342.  
  343.  
  344. ---
  345. home*CLI>
  346. <--- SIP read from UDP:198.65.166.131:5060 --->
  347. SIP/2.0 404 Not here
  348. Via: SIP/2.0/UDP 98.226.XXX.XXX:5060;branch=z9hG4bK4bc0cb5c;rport=5060
  349. From: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  350. To: <sip:+18008675309@10.218.63.140>;tag=1422683903
  351. Call-ID: 277698478@10.218.63.140
  352. CSeq: 102 BYE
  353. Server: OpenSIPS (1.6.2-notls (x86_64/linux))
  354. Content-Length: 0
  355. RemoteIP: 74.125.64.80
  356.  
  357.  
  358. <------------->
  359. --- (9 headers 0 lines) ---
  360. Really destroying SIP dialog '277698478@10.218.63.140' Method: ACK
  361. Really destroying SIP dialog '24257976032b21900b3a862e5b2e186e@192.168.2.250' Method: BYE
  362. home*CLI>
  363. <--- SIP read from UDP:198.65.166.131:5060 --->
  364. BYE sip:s@98.226.XXX.XXX:5060 SIP/2.0
  365. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  366. Record-Route: <sip:198.65.166.131;lr;ftag=1422683903>
  367. Call-ID: 277698478@10.218.63.140
  368. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  369. To: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  370. P-RTP-Stat: PS=547,OS=87520,PR=118,OR=18880,PL=0
  371. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK5922.30470286.0
  372. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK5922.20470286.0
  373. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK5922.ed1f45c7.0
  374. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK2051701148
  375. CSeq: 68663 BYE
  376. User-Agent: YATE/3.0.0
  377. Max-Forwards: 67
  378. Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
  379. Content-Length: 0
  380. RemoteIP: 74.125.64.80
  381. P-hint: rr-enforced
  382. P-hint: rr-enforced
  383.  
  384.  
  385. <------------->
  386. --- (19 headers 0 lines) ---
  387. home*CLI>
  388. <--- Transmitting (NAT) to 198.65.166.131:5060 --->
  389. SIP/2.0 481 Call leg/transaction does not exist
  390. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK5922.30470286.0;received=198.65.166.131
  391. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK5922.20470286.0
  392. Via: SIP/2.0/UDP 74.125.64.80:24000;branch=z9hG4bK5922.ed1f45c7.0
  393. Via: SIP/2.0/UDP 10.218.63.140:7654;received=10.218.63.140;rport=7654;branch=z9hG4bK2051701148
  394. From: <sip:+18008675309@10.218.63.140>;tag=1422683903
  395. To: <sip:747317XXXX@74.125.64.80:24000>;tag=as3de8e15e
  396. Call-ID: 277698478@10.218.63.140
  397. CSeq: 68663 BYE
  398. Server: Asterisk PBX 1.6.2.5-0ubuntu1
  399. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  400. Supported: replaces, timer
  401. Content-Length: 0
  402.  
  403.  
  404. <------------>
  405. [Sep 7 01:04:23] NOTICE[13480]: chan_sip.c:11461 sip_reregister: -- Re-registration for 1747317XXXX@proxy01.sipphone.com
  406. REGISTER 12 headers, 0 lines
  407. Reliably Transmitting (no NAT) to 198.65.166.131:5060:
  408. REGISTER sip:proxy01.sipphone.com SIP/2.0
  409. Via: SIP/2.0/UDP 98.226.XXX.XXX:5060;branch=z9hG4bK4baed1df;rport
  410. Max-Forwards: 70
  411. From: <sip:1747317XXXX@proxy01.sipphone.com>;tag=as044e0684
  412. To: <sip:1747317XXXX@proxy01.sipphone.com>
  413. Call-ID: 41d98fa00983743c2da5226e46cdb435@proxy01.sipphone.com
  414. CSeq: 117 REGISTER
  415. User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
  416. Authorization: Digest username="1747317XXXX", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4c85c86ddb286d53929d8ab69700c6d588ab24ea", response="ba3eec25020f5b819579cbc3522b4022"
  417. Expires: 120
  418. Contact: <sip:s@98.226.XXX.XXX>
  419. Content-Length: 0
  420.  
  421.  
  422. ---
  423. home*CLI> sip set debug off
  424. <--- SIP read from UDP:198.65.166.131:5060 --->
  425. SIP/2.0 200 OK
  426. Via: SIP/2.0/UDP 98.226.XXX.XXX:5060;branch=z9hG4bK4baed1df;rport=5060
  427. From: <sip:1747317XXXX@proxy01.sipphone.com>;tag=as044e0684
  428. To: <sip:1747317XXXX@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.4b4c
  429. Call-ID: 41d98fa00983743c2da5226e46cdb435@proxy01.sipphone.com
  430. CSeq: 117 REGISTER
  431. Contact: <sip:s@98.226.XXX.XXX>;expires=120
  432. Content-Length: 0
  433.  
  434.  
  435. <------------->
  436. --- (8 headers 0 lines) ---
  437. Scheduling destruction of SIP dialog '41d98fa00983743c2da5226e46cdb435@proxy01.sipphone.com' in 7936 ms (Method: REGISTER)
  438. [Sep 7 01:04:23] NOTICE[13480]: chan_sip.c:18179 handle_response_register: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s)
  439. home*CLI> sip set debug off
  440. SIP Debugging Disabled
  441. home*CLI>
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