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sip_debug.log

Mar 26th, 2014
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  1. Privilege escalation protection disabled!
  2. See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
  3. Asterisk 11.7.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  4. Created by Mark Spencer <markster@digium.com>
  5. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  6. This is free software, with components licensed under the GNU General Public
  7. License version 2 and other licenses; you are welcome to redistribute it under
  8. certain conditions. Type 'core show license' for details.
  9. =========================================================================
  10. Connected to Asterisk 11.7.0 currently running on LaneAsterix (pid = 1345)
  11. LaneAsterix*CLI>
  12.  == WebSocket connection from '10.168.107.7:62925' closed
  13.  
  14. LaneAsterix*CLI>
  15.  == WebSocket connection from '10.168.107.7:62924' closed
  16.  
  17. LaneAsterix*CLI>
  18.  == WebSocket connection from '10.168.107.7:62966' for protocol 'sip' accepted using version '13'
  19.  
  20. LaneAsterix*CLI>
  21. 
  22. <--- SIP read from WS:10.168.107.7:62966 --->
  23. REGISTER sip:10.168.100.160 SIP/2.0
  24. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKM6ctuk1qUGd1alQypTApHELQ5ZEDn4mm;rport
  25. From: "josh"<sip:josh@10.168.100.160>;tag=eBo7u7tR2REVmPPe3mAx
  26. To: "josh"<sip:josh@10.168.100.160>
  27. Contact: "josh"<sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  28. Call-ID: 2ba9cb4d-4881-1bb5-462f-8149e01bec0b
  29. CSeq: 47464 REGISTER
  30. Content-Length: 0
  31. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  32. Max-Forwards: 70
  33. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  34. Organization: Doubango Telecom
  35. Supported: path
  36.  
  37. <------------->
  38.  
  39. LaneAsterix*CLI>
  40. --- (13 headers 0 lines) ---
  41.  
  42. LaneAsterix*CLI>
  43. 
  44. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  45. SIP/2.0 401 Unauthorized
  46. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKM6ctuk1qUGd1alQypTApHELQ5ZEDn4mm;rport;received=10.168.107.7
  47. From: "josh"<sip:josh@10.168.100.160>;tag=eBo7u7tR2REVmPPe3mAx
  48. To: "josh"<sip:josh@10.168.100.160>;tag=as38754163
  49. Call-ID: 2ba9cb4d-4881-1bb5-462f-8149e01bec0b
  50. CSeq: 47464 REGISTER
  51. Server: Asterisk PBX 11.7.0
  52. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  53. Supported: replaces, timer
  54. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02be1a42"
  55. Content-Length: 0
  56.  
  57.  
  58. <------------>
  59.  
  60. LaneAsterix*CLI>
  61. Scheduling destruction of SIP dialog '2ba9cb4d-4881-1bb5-462f-8149e01bec0b' in 32000 ms (Method: REGISTER)
  62.  
  63. LaneAsterix*CLI>
  64. 
  65. <--- SIP read from WS:10.168.107.7:62966 --->
  66. REGISTER sip:10.168.100.160 SIP/2.0
  67. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkqte9GbLqw3pcfEenu1cEKN6Pvwkpu5S;rport
  68. From: "josh"<sip:josh@10.168.100.160>;tag=eBo7u7tR2REVmPPe3mAx
  69. To: "josh"<sip:josh@10.168.100.160>
  70. Contact: "josh"<sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  71. Call-ID: 2ba9cb4d-4881-1bb5-462f-8149e01bec0b
  72. CSeq: 47465 REGISTER
  73. Content-Length: 0
  74. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  75. Max-Forwards: 70
  76. Authorization: Digest username="josh",realm="asterisk",nonce="02be1a42",uri="sip:10.168.100.160",response="53419dd1165d9f13642a44fa0b06d80a",algorithm=MD5
  77. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  78. Organization: Doubango Telecom
  79. Supported: path
  80.  
  81. <------------->
  82.  
  83. LaneAsterix*CLI>
  84. --- (14 headers 0 lines) ---
  85.  
  86. LaneAsterix*CLI>
  87.  -- Registered SIP 'josh' at 10.168.107.7:62966
  88.  
  89. LaneAsterix*CLI>
  90. 
  91. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  92. SIP/2.0 200 OK
  93. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkqte9GbLqw3pcfEenu1cEKN6Pvwkpu5S;rport;received=10.168.107.7
  94. From: "josh"<sip:josh@10.168.100.160>;tag=eBo7u7tR2REVmPPe3mAx
  95. To: "josh"<sip:josh@10.168.100.160>;tag=as38754163
  96. Call-ID: 2ba9cb4d-4881-1bb5-462f-8149e01bec0b
  97. CSeq: 47465 REGISTER
  98. Server: Asterisk PBX 11.7.0
  99. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  100. Supported: replaces, timer
  101. Expires: 200
  102. Contact: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
  103. Date: Wed, 26 Mar 2014 23:32:36 GMT
  104. Content-Length: 0
  105.  
  106.  
  107. <------------>
  108.  
  109. LaneAsterix*CLI>
  110. Scheduling destruction of SIP dialog '2ba9cb4d-4881-1bb5-462f-8149e01bec0b' in 32000 ms (Method: REGISTER)
  111.  
  112. LaneAsterix*CLI>
  113.  == WebSocket connection from '10.168.107.7:62968' for protocol 'sip' accepted using version '13'
  114.  
  115. LaneAsterix*CLI>
  116. 
  117. <--- SIP read from WS:10.168.107.7:62968 --->
  118. REGISTER sip:10.168.100.160 SIP/2.0
  119. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKIhotman3zqfdGse1Cgk3Kf7u813qiruE;rport
  120. From: "tim"<sip:tim@10.168.100.160>;tag=LftsQGk75w8Z976JnuNH
  121. To: "tim"<sip:tim@10.168.100.160>
  122. Contact: "tim"<sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  123. Call-ID: c07ee691-2f11-fedb-c5c9-b674fa4e7f34
  124. CSeq: 48856 REGISTER
  125. Content-Length: 0
  126. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  127. Max-Forwards: 70
  128. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  129. Organization: Doubango Telecom
  130. Supported: path
  131.  
  132. <------------->
  133.  
  134. LaneAsterix*CLI>
  135. --- (13 headers 0 lines) ---
  136.  
  137. LaneAsterix*CLI>
  138. 
  139. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  140. SIP/2.0 401 Unauthorized
  141. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKIhotman3zqfdGse1Cgk3Kf7u813qiruE;rport;received=10.168.107.7
  142. From: "tim"<sip:tim@10.168.100.160>;tag=LftsQGk75w8Z976JnuNH
  143. To: "tim"<sip:tim@10.168.100.160>;tag=as442f7e8c
  144. Call-ID: c07ee691-2f11-fedb-c5c9-b674fa4e7f34
  145. CSeq: 48856 REGISTER
  146. Server: Asterisk PBX 11.7.0
  147. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  148. Supported: replaces, timer
  149. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2922b84e"
  150. Content-Length: 0
  151.  
  152.  
  153. <------------>
  154.  
  155. LaneAsterix*CLI>
  156. Scheduling destruction of SIP dialog 'c07ee691-2f11-fedb-c5c9-b674fa4e7f34' in 32000 ms (Method: REGISTER)
  157.  
  158. LaneAsterix*CLI>
  159. 
  160. <--- SIP read from WS:10.168.107.7:62968 --->
  161. REGISTER sip:10.168.100.160 SIP/2.0
  162. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK09TkFlzJ4fmCuhkx5bHGDnTuG4RO4WDp;rport
  163. From: "tim"<sip:tim@10.168.100.160>;tag=LftsQGk75w8Z976JnuNH
  164. To: "tim"<sip:tim@10.168.100.160>
  165. Contact: "tim"<sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  166. Call-ID: c07ee691-2f11-fedb-c5c9-b674fa4e7f34
  167. CSeq: 48857 REGISTER
  168. Content-Length: 0
  169. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  170. Max-Forwards: 70
  171. Authorization: Digest username="tim",realm="asterisk",nonce="2922b84e",uri="sip:10.168.100.160",response="3da82c49c5dff2c274f2d6a3de197939",algorithm=MD5
  172. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  173. Organization: Doubango Telecom
  174. Supported: path
  175.  
  176. <------------->
  177.  
  178. LaneAsterix*CLI>
  179. --- (14 headers 0 lines) ---
  180.  
  181. LaneAsterix*CLI>
  182.  -- Registered SIP 'tim' at 10.168.107.7:62968
  183.  
  184. LaneAsterix*CLI>
  185. 
  186. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  187. SIP/2.0 200 OK
  188. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK09TkFlzJ4fmCuhkx5bHGDnTuG4RO4WDp;rport;received=10.168.107.7
  189. From: "tim"<sip:tim@10.168.100.160>;tag=LftsQGk75w8Z976JnuNH
  190. To: "tim"<sip:tim@10.168.100.160>;tag=as442f7e8c
  191. Call-ID: c07ee691-2f11-fedb-c5c9-b674fa4e7f34
  192. CSeq: 48857 REGISTER
  193. Server: Asterisk PBX 11.7.0
  194. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  195. Supported: replaces, timer
  196. Expires: 200
  197. Contact: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
  198. Date: Wed, 26 Mar 2014 23:32:38 GMT
  199. Content-Length: 0
  200.  
  201.  
  202. <------------>
  203.  
  204. LaneAsterix*CLI>
  205. Scheduling destruction of SIP dialog 'c07ee691-2f11-fedb-c5c9-b674fa4e7f34' in 32000 ms (Method: REGISTER)
  206.  
  207. LaneAsterix*CLI>
  208. 
  209. <--- SIP read from WS:10.168.107.7:62966 --->
  210. INVITE sip:2002@10.168.100.160 SIP/2.0
  211. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB48QcpZahEhC29D8HuLuJGMUS6xFdvT4;rport
  212. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  213. To: <sip:2002@10.168.100.160>
  214. Contact: "josh"<sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=josh;ha1=be96c17aca7a7ff2d685d30341a5d42e;+g.oma.sip-im;+sip.ice;language="en,fr"
  215. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  216. CSeq: 35216 INVITE
  217. Content-Type: application/sdp
  218. Content-Length: 2902
  219. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  220. Max-Forwards: 70
  221. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  222. Organization: Doubango Telecom
  223.  
  224. v=0
  225. o=- 4338137477709027300 2 IN IP4 127.0.0.1
  226. s=Doubango Telecom - chrome
  227. t=0 0
  228. a=group:BUNDLE audio
  229. a=msid-semantic: WMS XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4
  230. m=audio 52437 RTP/SAVPF 111 103 104 0 8 106 105 13 126
  231. c=IN IP4 192.168.56.1
  232. a=rtcp:52437 IN IP4 192.168.56.1
  233. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 52437 typ host generation 0
  234. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 52437 typ host generation 0
  235. a=candidate:1418740086 1 udp 2113937151 192.168.1.143 52438 typ host generation 0
  236. a=candidate:1418740086 2 udp 2113937151 192.168.1.143 52438 typ host generation 0
  237. a=candidate:828646434 1 udp 2113937151 192.168.220.1 52439 typ host generation 0
  238. a=candidate:828646434 2 udp 2113937151 192.168.220.1 52439 typ host generation 0
  239. a=candidate:3172430149 1 udp 2113937151 10.168.107.7 52440 typ host generation 0
  240. a=candidate:3172430149 2 udp 2113937151 10.168.107.7 52440 typ host generation 0
  241. a=candidate:1448336772 1 udp 2113937151 192.168.182.1 52441 typ host generation 0
  242. a=candidate:1448336772 2 udp 2113937151 192.168.182.1 52441 typ host generation 0
  243. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
  244. a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
  245. a=candidate:437501318 1 tcp 1509957375 192.168.1.143 0 typ host generation 0
  246. a=candidate:437501318 2 tcp 1509957375 192.168.1.143 0 typ host generation 0
  247. a=candidate:2145900754 1 tcp 1509957375 192.168.220.1 0 typ host generation 0
  248. a=candidate:2145900754 2 tcp 1509957375 192.168.220.1 0 typ host generation 0
  249. a=candidate:4086599093 1 tcp 1509957375 10.168.107.7 0 typ host generation 0
  250. a=candidate:4086599093 2 tcp 1509957375 10.168.107.7 0 typ host generation 0
  251. a=candidate:416293236 1 tcp 1509957375 192.168.182.1 0 typ host generation 0
  252. a=candidate:416293236 2 tcp 1509957375 192.168.182.1 0 typ host generation 0
  253. a=ice-ufrag:NC8g4x9gyeZCOLcs
  254. a=ice-pwd:t/xJoLkkPtVZ/+wQNdNa9hvu
  255. a=ice-options:google-ice
  256. a=fingerprint:sha-256 F0:6D:1C:F5:93:4D:9F:EE:69:FA:34:42:AC:93:52:3D:26:13:87:CE:04:AF:D8:77:8B:83:5A:A2:57:2F:00:8F
  257. a=setup:actpass
  258. a=mid:audio
  259. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  260. a=sendrecv
  261. a=rtcp-mux
  262. a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:UmGqeev6B6FeN6V+kAZ4608ghxgjVXWS22pvqmfm
  263. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Pm5XMCFAgWc5wZrv3W+6z9QCVEINTI9N5FrdM7Rj
  264. a=rtpmap:111 opus/48000/2
  265. a=fmtp:111 minptime=10
  266. a=rtpmap:103 ISAC/16000
  267. a=rtpmap:104 ISAC/32000
  268. a=rtpmap:0 PCMU/8000
  269. a=rtpmap:8 PCMA/8000
  270. a=rtpmap:106 CN/32000
  271. a=rtpmap:105 CN/16000
  272. a=rtpmap:13 CN/8000
  273. a=rtpmap:126 telephone-event/8000
  274. a=maxptime:60
  275. a=ssrc:2582709043 cname:yriIHg3QolHWYSWq
  276. a=ssrc:2582709043 msid:XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4 67ba3f17-aa39-408b-88a5-8f5eeea2a58c
  277. a=ssrc:2582709043 mslabel:XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4
  278. a=ssrc:2582709043 label:67ba3f17-aa39-408b-88a5-8f5eeea2a58c
  279. <------------->
  280. --- (13 headers 55 lines) ---
  281. Using INVITE request as basis request - b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  282. Found peer 'josh' for 'josh' from 10.168.107.7:62966
  283.  
  284. <--- Reliably Transmitting (no NAT) to 10.168.107.7:5060 --->
  285. SIP/2.0 401 Unauthorized
  286. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB48QcpZahEhC29D8HuLuJGMUS6xFdvT4;rport;received=10.168.107.7
  287. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  288. To: <sip:2002@10.168.100.160>;tag=as12631c72
  289. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  290. CSeq: 35216 INVITE
  291. Server: Asterisk PBX 11.7.0
  292. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  293. Supported: replaces, timer
  294. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a4d94b2"
  295. Content-Length: 0
  296.  
  297.  
  298. <------------>
  299. Scheduling destruction of SIP dialog 'b29558c7-27c4-5f41-b372-6dc2adbb0fa2' in 32000 ms (Method: INVITE)
  300.  
  301. LaneAsterix*CLI>
  302. 
  303. <--- SIP read from WS:10.168.107.7:62966 --->
  304. ACK sip:2002@10.168.100.160 SIP/2.0
  305. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB48QcpZahEhC29D8HuLuJGMUS6xFdvT4;rport
  306. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  307. To: <sip:2002@10.168.100.160>;tag=as12631c72
  308. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  309. CSeq: 35216 ACK
  310. Content-Length: 0
  311. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  312. Max-Forwards: 70
  313.  
  314. <------------->
  315. --- (9 headers 0 lines) ---
  316.  
  317. LaneAsterix*CLI>
  318. 
  319. <--- SIP read from WS:10.168.107.7:62966 --->
  320. INVITE sip:2002@10.168.100.160 SIP/2.0
  321. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3rgyFOBpkqD1hIFiVA9rSuiGL1k74nJJ;rport
  322. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  323. To: <sip:2002@10.168.100.160>
  324. Contact: "josh"<sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=josh;ha1=be96c17aca7a7ff2d685d30341a5d42e;+g.oma.sip-im;+sip.ice;language="en,fr"
  325. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  326. CSeq: 35217 INVITE
  327. Content-Type: application/sdp
  328. Content-Length: 2902
  329. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  330. Max-Forwards: 70
  331. Authorization: Digest username="josh",realm="asterisk",nonce="5a4d94b2",uri="sip:2002@10.168.100.160",response="15f478a6d39d9e54d3a7799b9cb47084",algorithm=MD5
  332. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  333. Organization: Doubango Telecom
  334.  
  335. v=0
  336. o=- 4338137477709027300 2 IN IP4 127.0.0.1
  337. s=Doubango Telecom - chrome
  338. t=0 0
  339. a=group:BUNDLE audio
  340. a=msid-semantic: WMS XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4
  341. m=audio 52437 RTP/SAVPF 111 103 104 0 8 106 105 13 126
  342. c=IN IP4 192.168.56.1
  343. a=rtcp:52437 IN IP4 192.168.56.1
  344. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 52437 typ host generation 0
  345. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 52437 typ host generation 0
  346. a=candidate:1418740086 1 udp 2113937151 192.168.1.143 52438 typ host generation 0
  347. a=candidate:1418740086 2 udp 2113937151 192.168.1.143 52438 typ host generation 0
  348. a=candidate:828646434 1 udp 2113937151 192.168.220.1 52439 typ host generation 0
  349. a=candidate:828646434 2 udp 2113937151 192.168.220.1 52439 typ host generation 0
  350. a=candidate:3172430149 1 udp 2113937151 10.168.107.7 52440 typ host generation 0
  351. a=candidate:3172430149 2 udp 2113937151 10.168.107.7 52440 typ host generation 0
  352. a=candidate:1448336772 1 udp 2113937151 192.168.182.1 52441 typ host generation 0
  353. a=candidate:1448336772 2 udp 2113937151 192.168.182.1 52441 typ host generation 0
  354. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
  355. a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
  356. a=candidate:437501318 1 tcp 1509957375 192.168.1.143 0 typ host generation 0
  357. a=candidate:437501318 2 tcp 1509957375 192.168.1.143 0 typ host generation 0
  358. a=candidate:2145900754 1 tcp 1509957375 192.168.220.1 0 typ host generation 0
  359. a=candidate:2145900754 2 tcp 1509957375 192.168.220.1 0 typ host generation 0
  360. a=candidate:4086599093 1 tcp 1509957375 10.168.107.7 0 typ host generation 0
  361. a=candidate:4086599093 2 tcp 1509957375 10.168.107.7 0 typ host generation 0
  362. a=candidate:416293236 1 tcp 1509957375 192.168.182.1 0 typ host generation 0
  363. a=candidate:416293236 2 tcp 1509957375 192.168.182.1 0 typ host generation 0
  364. a=ice-ufrag:NC8g4x9gyeZCOLcs
  365. a=ice-pwd:t/xJoLkkPtVZ/+wQNdNa9hvu
  366. a=ice-options:google-ice
  367. a=fingerprint:sha-256 F0:6D:1C:F5:93:4D:9F:EE:69:FA:34:42:AC:93:52:3D:26:13:87:CE:04:AF:D8:77:8B:83:5A:A2:57:2F:00:8F
  368. a=setup:actpass
  369. a=mid:audio
  370. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  371. a=sendrecv
  372. a=rtcp-mux
  373. a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:UmGqeev6B6FeN6V+kAZ4608ghxgjVXWS22pvqmfm
  374. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Pm5XMCFAgWc5wZrv3W+6z9QCVEINTI9N5FrdM7Rj
  375. a=rtpmap:111 opus/48000/2
  376. a=fmtp:111 minptime=10
  377. a=rtpmap:103 ISAC/16000
  378. a=rtpmap:104 ISAC/32000
  379. a=rtpmap:0 PCMU/8000
  380. a=rtpmap:8 PCMA/8000
  381. a=rtpmap:106 CN/32000
  382. a=rtpmap:105 CN/16000
  383. a=rtpmap:13 CN/8000
  384. a=rtpmap:126 telephone-event/8000
  385. a=maxptime:60
  386. a=ssrc:2582709043 cname:yriIHg3QolHWYSWq
  387. a=ssrc:2582709043 msid:XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4 67ba3f17-aa39-408b-88a5-8f5eeea2a58c
  388. a=ssrc:2582709043 mslabel:XxfnT1yMA0ilvpHK9TTTXi1iPSpklbFmLjh4
  389. a=ssrc:2582709043 label:67ba3f17-aa39-408b-88a5-8f5eeea2a58c
  390. <------------->
  391. --- (14 headers 55 lines) ---
  392. Using INVITE request as basis request - b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  393. Found peer 'josh' for 'josh' from 10.168.107.7:62966
  394. == Using SIP RTP CoS mark 5
  395. [Mar 26 19:32:46] NOTICE[1699][C-00000004]: chan_sip.c:10114 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 52437 RTP/SAVPF 111 103 104 0 8 106 105 13 126
  396. Found RTP audio format 111
  397. Found RTP audio format 103
  398. Found RTP audio format 104
  399. Found RTP audio format 0
  400. Found RTP audio format 8
  401. Found RTP audio format 106
  402. Found RTP audio format 105
  403. Found RTP audio format 13
  404. Found RTP audio format 126
  405. Found unknown media description format opus for ID 111
  406. Found unknown media description format ISAC for ID 103
  407. Found unknown media description format ISAC for ID 104
  408. Found audio description format PCMU for ID 0
  409. Found audio description format PCMA for ID 8
  410. Found unknown media description format CN for ID 106
  411. Found unknown media description format CN for ID 105
  412. Found audio description format CN for ID 13
  413. Found audio description format telephone-event for ID 126
  414. Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  415. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  416. Peer audio RTP is at port 192.168.56.1:52437
  417. Looking for 2002 in users (domain 10.168.100.160)
  418. list_route: hop: <sip:josh@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>
  419.  
  420. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  421. SIP/2.0 100 Trying
  422. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3rgyFOBpkqD1hIFiVA9rSuiGL1k74nJJ;rport;received=10.168.107.7
  423. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  424. To: <sip:2002@10.168.100.160>
  425. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  426. CSeq: 35217 INVITE
  427. Server: Asterisk PBX 11.7.0
  428. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  429. Supported: replaces, timer
  430. Contact: <sip:2002@10.168.100.160:5060;transport=WS>
  431. Content-Length: 0
  432.  
  433.  
  434. <------------>
  435.  
  436. LaneAsterix*CLI>
  437.  -- Executing [2002@users:1] Dial("SIP/josh-00000008", "SIP/tim,15") in new stack
  438.  
  439. LaneAsterix*CLI>
  440.  == Using SIP RTP CoS mark 5
  441.  
  442. LaneAsterix*CLI>
  443. [Mar 26 19:32:46] ERROR[1701][C-00000004]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Temporary failure in name resolution
  444.  
  445. LaneAsterix*CLI>
  446. [Mar 26 19:32:46] WARNING[1701][C-00000004]: chan_sip.c:15881 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  447.  
  448. LaneAsterix*CLI>
  449. [Mar 26 19:32:46] ERROR[1701][C-00000004]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
  450.  
  451. LaneAsterix*CLI>
  452. Audio is at 12272
  453.  
  454. LaneAsterix*CLI>
  455. Adding codec 100003 (ulaw) to SDP
  456.  
  457. LaneAsterix*CLI>
  458. Adding codec 100002 (gsm) to SDP
  459.  
  460. LaneAsterix*CLI>
  461. Adding codec 100004 (alaw) to SDP
  462.  
  463. LaneAsterix*CLI>
  464. Adding codec 100017 (testlaw) to SDP
  465.  
  466. LaneAsterix*CLI>
  467. Adding non-codec 0x1 (telephone-event) to SDP
  468.  
  469. LaneAsterix*CLI>
  470. Reliably Transmitting (NAT) to 10.168.107.7:62968:
  471. INVITE sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
  472. Via: SIP/2.0/WS 10.168.100.160:5060;branch=z9hG4bK35ce0bbe;rport
  473. Max-Forwards: 70
  474. From: "Josh" <sip:2001@10.168.100.160>;tag=as26de3f0c
  475. To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  476. Contact: <sip:2001@10.168.100.160:5060;transport=WS>
  477. Call-ID: 004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060
  478. CSeq: 102 INVITE
  479. User-Agent: Asterisk PBX 11.7.0
  480. Date: Wed, 26 Mar 2014 23:32:46 GMT
  481. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  482. Supported: replaces, timer
  483. Content-Type: application/sdp
  484. Content-Length: 599
  485.  
  486. v=0
  487. o=root 1287776921 1287776921 IN IP4 10.168.100.160
  488. s=Asterisk PBX 11.7.0
  489. c=IN IP4 10.168.100.160
  490. t=0 0
  491. m=audio 12272 RTP/SAVP 0 3 8 101
  492. a=rtpmap:0 PCMU/8000
  493. a=rtpmap:3 GSM/8000
  494. a=rtpmap:8 PCMA/8000
  495. a=rtpmap:101 telephone-event/8000
  496. a=fmtp:101 0-16
  497. a=ptime:20
  498. a=ice-ufrag:07d8856e53600eb720d438de52779356
  499. a=ice-pwd:1361717b2dd224472fd617ac6938e745
  500. a=candidate:Haa864a0 1 UDP 2130706431 10.168.100.160 12272 typ host
  501. a=candidate:Haa864a0 2 UDP 2130706430 10.168.100.160 12273 typ host
  502. a=sendrecv
  503. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CkGLkRK7xoXbc3iM6pvv6aTOtem9nhLsRBLpQcqY
  504.  
  505. ---
  506.  
  507. LaneAsterix*CLI>
  508.  -- Called SIP/tim
  509.  
  510. LaneAsterix*CLI>
  511. 
  512. <--- SIP read from WS:10.168.107.7:62968 --->
  513. SIP/2.0 100 Trying (sent from the Transaction Layer)
  514. Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK35ce0bbe
  515. From: "Josh"<sip:2001@10.168.100.160>;tag=as26de3f0c
  516. To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  517. Call-ID: 004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060
  518. CSeq: 102 INVITE
  519. Content-Length: 0
  520.  
  521. <------------->
  522. --- (7 headers 0 lines) ---
  523.  
  524. LaneAsterix*CLI>
  525. 
  526. <--- SIP read from WS:10.168.107.7:62968 --->
  527. SIP/2.0 180 Ringing
  528. Via: SIP/2.0/WS 10.168.100.160:5060;rport=5060;branch=z9hG4bK35ce0bbe
  529. From: "Josh"<sip:2001@10.168.100.160>;tag=as26de3f0c
  530. To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=1sSAgg1O4Fuw1ybFDQei
  531. Contact: <sip:tim@df7jal23ls0d.invalid;transport=ws>
  532. Call-ID: 004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060
  533. CSeq: 102 INVITE
  534. Content-Length: 0
  535. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  536.  
  537. <------------->
  538. --- (9 headers 0 lines) ---
  539. list_route: hop: <sip:tim@df7jal23ls0d.invalid;transport=ws>
  540.  
  541. LaneAsterix*CLI>
  542.  -- SIP/tim-00000009 is ringing
  543.  
  544. LaneAsterix*CLI>
  545. 
  546. <--- Transmitting (no NAT) to 10.168.107.7:5060 --->
  547. SIP/2.0 180 Ringing
  548. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3rgyFOBpkqD1hIFiVA9rSuiGL1k74nJJ;rport;received=10.168.107.7
  549. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  550. To: <sip:2002@10.168.100.160>;tag=as412b3bf8
  551. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  552. CSeq: 35217 INVITE
  553. Server: Asterisk PBX 11.7.0
  554. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  555. Supported: replaces, timer
  556. Contact: <sip:2002@10.168.100.160:5060;transport=WS>
  557. Content-Length: 0
  558.  
  559.  
  560. <------------>
  561.  
  562. LaneAsterix*CLI>
  563.  -- Nobody picked up in 15000 ms
  564. Scheduling destruction of SIP dialog '004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060' in 32000 ms (Method: INVITE)
  565. Reliably Transmitting (NAT) to 10.168.107.7:62968:
  566. CANCEL sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
  567. Via: SIP/2.0/WS 10.168.100.160:5060;branch=z9hG4bK35ce0bbe;rport
  568. Max-Forwards: 70
  569. From: "Josh" <sip:2001@10.168.100.160>;tag=as26de3f0c
  570. To: <sip:tim@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  571. Call-ID: 004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060
  572. CSeq: 102 CANCEL
  573. User-Agent: Asterisk PBX 11.7.0
  574. Content-Length: 0
  575.  
  576.  
  577. ---
  578. Scheduling destruction of SIP dialog '004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060' in 32000 ms (Method: INVITE)
  579. -- Auto fallthrough, channel 'SIP/josh-00000008' status is 'NOANSWER'
  580. Scheduling destruction of SIP dialog 'b29558c7-27c4-5f41-b372-6dc2adbb0fa2' in 32000 ms (Method: INVITE)
  581.  
  582. <--- Reliably Transmitting (no NAT) to 10.168.107.7:5060 --->
  583. SIP/2.0 603 Declined
  584. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3rgyFOBpkqD1hIFiVA9rSuiGL1k74nJJ;rport;received=10.168.107.7
  585. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  586. To: <sip:2002@10.168.100.160>;tag=as412b3bf8
  587. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  588. CSeq: 35217 INVITE
  589. Server: Asterisk PBX 11.7.0
  590. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  591. Supported: replaces, timer
  592. Content-Length: 0
  593.  
  594.  
  595. <------------>
  596.  
  597. LaneAsterix*CLI>
  598. 
  599. <--- SIP read from WS:10.168.107.7:62966 --->
  600. ACK sip:2002@10.168.100.160 SIP/2.0
  601. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3rgyFOBpkqD1hIFiVA9rSuiGL1k74nJJ;rport
  602. From: "josh"<sip:josh@10.168.100.160>;tag=kLohP63xBVN0yElLDt8R
  603. To: <sip:2002@10.168.100.160>;tag=as412b3bf8
  604. Call-ID: b29558c7-27c4-5f41-b372-6dc2adbb0fa2
  605. CSeq: 35217 ACK
  606. Content-Length: 0
  607. Route: <sip:10.168.100.160:5060;lr;sipml5-outbound;transport=udp>
  608. Max-Forwards: 70
  609.  
  610. <------------->
  611. --- (9 headers 0 lines) ---
  612.  
  613. LaneAsterix*CLI>
  614. Really destroying SIP dialog '2ba9cb4d-4881-1bb5-462f-8149e01bec0b' Method: REGISTER
  615.  
  616. LaneAsterix*CLI>
  617. Really destroying SIP dialog 'c07ee691-2f11-fedb-c5c9-b674fa4e7f34' Method: REGISTER
  618.  
  619. LaneAsterix*CLI>
  620. Really destroying SIP dialog '004d49ca5cc8bb712a4eeeb71ef49905@10.168.100.160:5060' Method: INVITE
  621. Really destroying SIP dialog 'b29558c7-27c4-5f41-b372-6dc2adbb0fa2' Method: INVITE
  622.  
  623. LaneAsterix*CLI>
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