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  1. SIP Debugging enabled
  2.   == WebSocket connection from '192.168.88.174:49537' for protocol 'sip' accepted using version '13'
  3.  
  4. <--- SIP read from WS:192.168.88.174:49537 --->
  5. REGISTER sip:192.168.88.251 SIP/2.0
  6. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport
  7. From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
  8. To: "888"<sip:888@192.168.88.251>
  9. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  10. Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
  11. CSeq: 6286 REGISTER
  12. Content-Length: 0
  13. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  14. Max-Forwards: 70
  15. Authorization: Digest username="888",realm="192.168.88.251",nonce="",uri="sip:192.168.88.251",response=""
  16. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  17. Organization: Doubango Telecom
  18. Supported: path
  19.  
  20. <------------->
  21. --- (14 headers 0 lines) ---
  22.  
  23. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  24. SIP/2.0 401 Unauthorized
  25. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport;received=192.168.88.174
  26. From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
  27. To: "888"<sip:888@192.168.88.251>;tag=as7404acc2
  28. Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
  29. CSeq: 6286 REGISTER
  30. Server: Asterisk PBX 13.2.0
  31. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  32. Supported: replaces, timer
  33. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="09171086"
  34. Content-Length: 0
  35.  
  36.  
  37. <------------>
  38. Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)
  39.  
  40. <--- SIP read from WS:192.168.88.174:49537 --->
  41. REGISTER sip:192.168.88.251 SIP/2.0
  42. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport
  43. From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
  44. To: "888"<sip:888@192.168.88.251>
  45. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  46. Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
  47. CSeq: 6287 REGISTER
  48. Content-Length: 0
  49. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  50. Max-Forwards: 70
  51. Authorization: Digest username="888",realm="192.168.88.251",nonce="09171086",uri="sip:192.168.88.251",response="7392735b12a3d08cee2d5aa586a8f225",algorithm=MD5
  52. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  53. Organization: Doubango Telecom
  54. Supported: path
  55.  
  56. <------------->
  57. --- (14 headers 0 lines) ---
  58.     -- Registered SIP '888' at 192.168.88.174:49537
  59.  
  60. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  61. SIP/2.0 200 OK
  62. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport;received=192.168.88.174
  63. From: "888"<sip:888@192.168.88.251>;tag=nnfL8hLlvU1QirQeN3rT
  64. To: "888"<sip:888@192.168.88.251>;tag=as7404acc2
  65. Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
  66. CSeq: 6287 REGISTER
  67. Server: Asterisk PBX 13.2.0
  68. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  69. Supported: replaces, timer
  70. Expires: 200
  71. Contact: <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
  72. Date: Fri, 13 Feb 2015 04:15:38 GMT
  73. Content-Length: 0
  74.  
  75.  
  76. <------------>
  77. Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)
  78.  
  79. <--- SIP read from WS:192.168.88.174:49537 --->
  80. INVITE sip:889@192.168.88.251 SIP/2.0
  81. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
  82. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  83. To: <sip:889@192.168.88.251>
  84. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
  85. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  86. CSeq: 26632 INVITE
  87. Content-Type: application/sdp
  88. Content-Length: 1590
  89. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  90. Max-Forwards: 70
  91. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  92. Organization: Doubango Telecom
  93.  
  94. v=0
  95. o=- 437287842917936700 2 IN IP4 127.0.0.1
  96. s=Doubango Telecom - chrome
  97. t=0 0
  98. a=group:BUNDLE audio
  99. a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
  100. m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  101. c=IN IP4 192.168.88.174
  102. a=rtcp:65003 IN IP4 192.168.88.174
  103. a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
  104. a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
  105. a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  106. a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  107. a=ice-ufrag:E3Z2rHZDSTilnu0z
  108. a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
  109. a=ice-options:google-ice
  110. a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
  111. a=setup:actpass
  112. a=mid:audio
  113. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  114. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  115. a=sendrecv
  116. a=rtcp-mux
  117. a=rtpmap:111 opus/48000/2
  118. a=fmtp:111 minptime=10
  119. a=rtpmap:103 ISAC/16000
  120. a=rtpmap:104 ISAC/32000
  121. a=rtpmap:9 G722/8000
  122. a=rtpmap:0 PCMU/8000
  123. a=rtpmap:8 PCMA/8000
  124. a=rtpmap:106 CN/32000
  125. a=rtpmap:105 CN/16000
  126. a=rtpmap:13 CN/8000
  127. a=rtpmap:126 telephone-event/8000
  128. a=maxptime:60
  129. a=ssrc:3257448580 cname:gVHxZ+t4jTqxgc7F
  130. a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
  131. a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
  132. a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
  133. <------------->
  134. --- (13 headers 39 lines) ---
  135. Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
  136. Found peer '888' for '888' from 192.168.88.174:49537
  137.  
  138. <--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
  139. SIP/2.0 401 Unauthorized
  140. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport;received=192.168.88.174
  141. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  142. To: <sip:889@192.168.88.251>;tag=as29afbf84
  143. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  144. CSeq: 26632 INVITE
  145. Server: Asterisk PBX 13.2.0
  146. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  147. Supported: replaces, timer
  148. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="5a02570a"
  149. Content-Length: 0
  150.  
  151.  
  152. <------------>
  153. Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)
  154.  
  155. <--- SIP read from WS:192.168.88.174:49537 --->
  156. ACK sip:889@192.168.88.251 SIP/2.0
  157. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
  158. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  159. To: <sip:889@192.168.88.251>;tag=as29afbf84
  160. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  161. CSeq: 26632 ACK
  162. Content-Length: 0
  163. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  164. Max-Forwards: 70
  165.  
  166. <------------->
  167. --- (9 headers 0 lines) ---
  168.  
  169. <--- SIP read from WS:192.168.88.174:49537 --->
  170. INVITE sip:889@192.168.88.251 SIP/2.0
  171. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport
  172. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  173. To: <sip:889@192.168.88.251>
  174. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
  175. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  176. CSeq: 26633 INVITE
  177. Content-Type: application/sdp
  178. Content-Length: 1590
  179. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  180. Max-Forwards: 70
  181. Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:889@192.168.88.251",response="d328cce01e71bf1f662eae00a41ed1aa",algorithm=MD5
  182. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  183. Organization: Doubango Telecom
  184.  
  185. v=0
  186. o=- 437287842917936700 2 IN IP4 127.0.0.1
  187. s=Doubango Telecom - chrome
  188. t=0 0
  189. a=group:BUNDLE audio
  190. a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
  191. m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  192. c=IN IP4 192.168.88.174
  193. a=rtcp:65003 IN IP4 192.168.88.174
  194. a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
  195. a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
  196. a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  197. a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  198. a=ice-ufrag:E3Z2rHZDSTilnu0z
  199. a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
  200. a=ice-options:google-ice
  201. a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
  202. a=setup:actpass
  203. a=mid:audio
  204. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  205. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  206. a=sendrecv
  207. a=rtcp-mux
  208. a=rtpmap:111 opus/48000/2
  209. a=fmtp:111 minptime=10
  210. a=rtpmap:103 ISAC/16000
  211. a=rtpmap:104 ISAC/32000
  212. a=rtpmap:9 G722/8000
  213. a=rtpmap:0 PCMU/8000
  214. a=rtpmap:8 PCMA/8000
  215. a=rtpmap:106 CN/32000
  216. a=rtpmap:105 CN/16000
  217. a=rtpmap:13 CN/8000
  218. a=rtpmap:126 telephone-event/8000
  219. a=maxptime:60
  220. a=ssrc:3257448580 cname:gVHxZ+t4jTqxgc7F
  221. a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
  222. a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
  223. a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
  224. <------------->
  225. --- (14 headers 39 lines) ---
  226. Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
  227. Found peer '888' for '888' from 192.168.88.174:49537
  228.   == Using SIP RTP CoS mark 5
  229. Found RTP audio format 111
  230. Found RTP audio format 103
  231. Found RTP audio format 104
  232. Found RTP audio format 9
  233. Found RTP audio format 0
  234. Found RTP audio format 8
  235. Found RTP audio format 106
  236. Found RTP audio format 105
  237. Found RTP audio format 13
  238. Found RTP audio format 126
  239. Found audio description format opus for ID 111
  240. Found unknown media description format ISAC for ID 103
  241. Found unknown media description format ISAC for ID 104
  242. Found audio description format G722 for ID 9
  243. Found audio description format PCMU for ID 0
  244. Found audio description format PCMA for ID 8
  245. Found unknown media description format CN for ID 106
  246. Found unknown media description format CN for ID 105
  247. Found audio description format CN for ID 13
  248. Found audio description format telephone-event for ID 126
  249. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  250. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  251. Peer audio RTP is at port 192.168.88.174:65003
  252. Looking for 889 in default (domain 192.168.88.251)
  253. sip_route_dump: route/path hop: <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>
  254.  
  255. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  256. SIP/2.0 100 Trying
  257. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
  258. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  259. To: <sip:889@192.168.88.251>
  260. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  261. CSeq: 26633 INVITE
  262. Server: Asterisk PBX 13.2.0
  263. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  264. Supported: replaces, timer
  265. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  266. Content-Length: 0
  267.  
  268.  
  269. <------------>
  270.     -- Executing [889@default:1] Dial("SIP/888-0000005f", "SIP/889") in new stack
  271.   == Using SIP RTP CoS mark 5
  272. Audio is at 13566
  273. Adding codec ulaw to SDP
  274. Adding codec alaw to SDP
  275. Adding codec gsm to SDP
  276. Adding non-codec 0x1 (telephone-event) to SDP
  277. Reliably Transmitting (no NAT) to 192.168.88.187:49625:
  278. INVITE sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
  279. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
  280. Max-Forwards: 70
  281. From: "888" <sip:888@192.168.88.251>;tag=as176937a0
  282. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  283. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  284. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  285. CSeq: 102 INVITE
  286. User-Agent: Asterisk PBX 13.2.0
  287. Date: Fri, 13 Feb 2015 04:15:46 GMT
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  289. Supported: replaces, timer
  290. Content-Type: application/sdp
  291. Content-Length: 676
  292.  
  293. v=0
  294. o=root 1187332515 1187332515 IN IP4 192.168.88.251
  295. s=Asterisk PBX 13.2.0
  296. c=IN IP4 192.168.88.251
  297. t=0 0
  298. m=audio 13566 RTP/SAVPF 0 8 3 101
  299. a=rtpmap:0 PCMU/8000
  300. a=rtpmap:8 PCMA/8000
  301. a=rtpmap:3 GSM/8000
  302. a=rtpmap:101 telephone-event/8000
  303. a=fmtp:101 0-16
  304. a=maxptime:150
  305. a=ice-ufrag:2870a6d64720723e773aa0364395d0f4
  306. a=ice-pwd:1c240a294ddc471a2b5ba61158281a7d
  307. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 13566 typ host
  308. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 13567 typ host
  309. a=connection:new
  310. a=setup:actpass
  311. a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
  312. a=sendrecv
  313.  
  314. ---
  315.     -- Called SIP/889
  316.  
  317. <--- SIP read from WS:192.168.88.187:49625 --->
  318. SIP/2.0 100 Trying (sent from the Transaction Layer)
  319. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
  320. From: "888"<sip:888@192.168.88.251>;tag=as176937a0
  321. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  322. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  323. CSeq: 102 INVITE
  324. Content-Length: 0
  325.  
  326. <------------->
  327. --- (7 headers 0 lines) ---
  328.  
  329. <--- SIP read from WS:192.168.88.187:49625 --->
  330. SIP/2.0 180 Ringing
  331. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
  332. From: "888"<sip:888@192.168.88.251>;tag=as176937a0
  333. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
  334. Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
  335. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  336. CSeq: 102 INVITE
  337. Content-Length: 0
  338. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  339.  
  340. <------------->
  341. --- (9 headers 0 lines) ---
  342. sip_route_dump: route/path hop: <sip:889@df7jal23ls0d.invalid;transport=ws>
  343.     -- SIP/889-00000060 is ringing
  344.  
  345. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  346. SIP/2.0 180 Ringing
  347. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
  348. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  349. To: <sip:889@192.168.88.251>;tag=as4947f672
  350. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  351. CSeq: 26633 INVITE
  352. Server: Asterisk PBX 13.2.0
  353. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  354. Supported: replaces, timer
  355. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  356. Content-Length: 0
  357.  
  358.  
  359. <------------>
  360.  
  361. <--- SIP read from WS:192.168.88.187:49625 --->
  362. REGISTER sip:192.168.88.251 SIP/2.0
  363. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport
  364. From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
  365. To: "889"<sip:889@192.168.88.251>
  366. Contact: "889"<sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  367. Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
  368. CSeq: 30884 REGISTER
  369. Content-Length: 0
  370. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  371. Max-Forwards: 70
  372. Authorization: Digest username="889",realm="192.168.88.251",nonce="094b5901",uri="sip:192.168.88.251",response="a635f03607d21873ed8a4a74975b97cc",algorithm=MD5
  373. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  374. Organization: Doubango Telecom
  375.  
  376. <------------->
  377. --- (13 headers 0 lines) ---
  378.  
  379. <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
  380. SIP/2.0 401 Unauthorized
  381. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport;received=192.168.88.187
  382. From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
  383. To: "889"<sip:889@192.168.88.251>;tag=as0ddfbb67
  384. Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
  385. CSeq: 30884 REGISTER
  386. Server: Asterisk PBX 13.2.0
  387. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  388. Supported: replaces, timer
  389. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="13fe51c6"
  390. Content-Length: 0
  391.  
  392.  
  393. <------------>
  394. Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)
  395.  
  396. <--- SIP read from WS:192.168.88.187:49625 --->
  397. REGISTER sip:192.168.88.251 SIP/2.0
  398. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport
  399. From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
  400. To: "889"<sip:889@192.168.88.251>
  401. Contact: "889"<sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  402. Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
  403. CSeq: 30885 REGISTER
  404. Content-Length: 0
  405. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  406. Max-Forwards: 70
  407. Authorization: Digest username="889",realm="192.168.88.251",nonce="13fe51c6",uri="sip:192.168.88.251",response="a1a4fccb04ffa5cef37a37267ce4cc4e",algorithm=MD5
  408. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  409. Organization: Doubango Telecom
  410.  
  411. <------------->
  412. --- (13 headers 0 lines) ---
  413.  
  414. <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
  415. SIP/2.0 200 OK
  416. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport;received=192.168.88.187
  417. From: "889"<sip:889@192.168.88.251>;tag=V0GF27XnCoBLdq6gumel
  418. To: "889"<sip:889@192.168.88.251>;tag=as0ddfbb67
  419. Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
  420. CSeq: 30885 REGISTER
  421. Server: Asterisk PBX 13.2.0
  422. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  423. Supported: replaces, timer
  424. Expires: 200
  425. Contact: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
  426. Date: Fri, 13 Feb 2015 04:15:49 GMT
  427. Content-Length: 0
  428.  
  429.  
  430. <------------>
  431. Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)
  432. Really destroying SIP dialog 'd49645e7-984f-10b5-863d-bb5631bf9285' Method: REGISTER
  433.  
  434. <--- SIP read from WS:192.168.88.187:49625 --->
  435. SIP/2.0 200 OK
  436. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
  437. From: "888"<sip:888@192.168.88.251>;tag=as176937a0
  438. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
  439. Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
  440. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  441. CSeq: 102 INVITE
  442. Content-Type: application/sdp
  443. Content-Length: 1171
  444. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  445.  
  446. v=0
  447. o=- 7710960356793537000 2 IN IP4 127.0.0.1
  448. s=Doubango Telecom - chrome
  449. t=0 0
  450. a=msid-semantic: WMS 85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
  451. m=audio 50026 UDP/TLS/RTP/SAVPF 0 8 101
  452. c=IN IP4 192.168.88.187
  453. a=rtcp:50027 IN IP4 192.168.88.187
  454. a=candidate:2577307183 1 udp 2122194687 192.168.88.187 50026 typ host generation 0
  455. a=candidate:2577307183 2 udp 2122194686 192.168.88.187 50027 typ host generation 0
  456. a=candidate:3609029343 1 tcp 1518214911 192.168.88.187 0 typ host tcptype active generation 0
  457. a=candidate:3609029343 2 tcp 1518214910 192.168.88.187 0 typ host tcptype active generation 0
  458. a=ice-ufrag:awyAyzrZujG0M9zq
  459. a=ice-pwd:z7uMvoajOVqtt3hHg+yAhqmj
  460. a=fingerprint:sha-256 D8:C4:BF:59:B9:A8:19:A0:4C:31:BA:92:F0:62:A0:3E:27:D4:90:9B:79:33:E3:B6:FC:E9:2A:EB:C3:D3:DF:E6
  461. a=setup:active
  462. a=mid:audio
  463. a=sendrecv
  464. a=rtpmap:0 PCMU/8000
  465. a=rtpmap:8 PCMA/8000
  466. a=rtpmap:101 telephone-event/8000
  467. a=ssrc:372692495 cname:pkv1+OHhG1pbR11N
  468. a=ssrc:372692495 msid:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO 4890e5af-7a2d-4d88-a39f-32216cbc0d0d
  469. a=ssrc:372692495 mslabel:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
  470. a=ssrc:372692495 label:4890e5af-7a2d-4d88-a39f-32216cbc0d0d
  471. <------------->
  472. --- (10 headers 25 lines) ---
  473. Found RTP audio format 0
  474. Found RTP audio format 8
  475. Found RTP audio format 101
  476. Found audio description format PCMU for ID 0
  477. Found audio description format PCMA for ID 8
  478. Found audio description format telephone-event for ID 101
  479. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  480. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  481. Peer audio RTP is at port 192.168.88.187:50026
  482. sip_route_dump: route/path hop: <sip:889@df7jal23ls0d.invalid;transport=ws>
  483. [Feb 13 06:15:59] ERROR[1055][C-00000031]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  484. [Feb 13 06:15:59] WARNING[1055][C-00000031]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  485. set_destination: Parsing <sip:889@df7jal23ls0d.invalid;transport=ws> for address/port to send to
  486. set_destination: URI is for WebSocket, we can't set destination
  487. Transmitting (no NAT) to 192.168.88.187:49625:
  488. ACK sip:889@df7jal23ls0d.invalid;transport=ws SIP/2.0
  489. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3a7ff3e4
  490. Max-Forwards: 70
  491. From: "888" <sip:888@192.168.88.251>;tag=as176937a0
  492. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
  493. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  494. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  495. CSeq: 102 ACK
  496. User-Agent: Asterisk PBX 13.2.0
  497. Content-Length: 0
  498.  
  499.  
  500. ---
  501.     -- SIP/889-00000060 answered SIP/888-0000005f
  502. Audio is at 16190
  503. Adding codec ulaw to SDP
  504. Adding codec alaw to SDP
  505. Adding codec gsm to SDP
  506. Adding non-codec 0x1 (telephone-event) to SDP
  507.  
  508. <--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
  509. SIP/2.0 200 OK
  510. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
  511. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  512. To: <sip:889@192.168.88.251>;tag=as4947f672
  513. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  514. CSeq: 26633 INVITE
  515. Server: Asterisk PBX 13.2.0
  516. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  517. Supported: replaces, timer
  518. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  519. Content-Type: application/sdp
  520. Content-Length: 673
  521.  
  522. v=0
  523. o=root 380639406 380639406 IN IP4 192.168.88.251
  524. s=Asterisk PBX 13.2.0
  525. c=IN IP4 192.168.88.251
  526. t=0 0
  527. m=audio 16190 RTP/SAVPF 0 8 3 126
  528. a=rtpmap:0 PCMU/8000
  529. a=rtpmap:8 PCMA/8000
  530. a=rtpmap:3 GSM/8000
  531. a=rtpmap:126 telephone-event/8000
  532. a=fmtp:126 0-16
  533. a=maxptime:150
  534. a=ice-ufrag:5c8047ce65b257e40fd0093a148878ce
  535. a=ice-pwd:4988ef1c009a1c717d3e1c2444e686ff
  536. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 16190 typ host
  537. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 16191 typ host
  538. a=connection:new
  539. a=setup:active
  540. a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
  541. a=sendrecv
  542.  
  543. <------------>
  544.     -- Channel SIP/888-0000005f joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
  545.     -- Channel SIP/889-00000060 joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
  546.  
  547. <--- SIP read from WS:192.168.88.174:49537 --->
  548. ACK sip:889@192.168.88.251:5060;transport=WS SIP/2.0
  549. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKClVdt1n8PFKSu96LXuzf;rport
  550. From: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  551. To: <sip:889@192.168.88.251>;tag=as4947f672
  552. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  553. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  554. CSeq: 26633 ACK
  555. Content-Length: 0
  556. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  557. Max-Forwards: 70
  558. Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:889@192.168.88.251:5060;transport=WS",response="780d8ba6cec222db3fac9c9f8de84e87",algorithm=MD5
  559. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  560. Organization: Doubango Telecom
  561.  
  562. <------------->
  563. --- (13 headers 0 lines) ---
  564.        > 0x7fd8389ed7e0 -- Probation passed - setting RTP source address to 192.168.88.187:50026
  565.  
  566. <--- SIP read from WS:192.168.88.187:49625 --->
  567. BYE sip:888@192.168.88.251:5060;transport=WS SIP/2.0
  568. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport
  569. From: <sip:889@df7jal23ls0d.invalid>;tag=7NfJJeWFCMBeSuhOFJ3T
  570. To: "888"<sip:888@192.168.88.251>;tag=as176937a0
  571. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  572. CSeq: 43548 BYE
  573. Content-Length: 0
  574. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  575. Max-Forwards: 70
  576. Accept-Contact: *;+g.oma.sip-im
  577. Accept-Contact: *;language="en,fr"
  578. Accept-Contact: *;+g.oma.sip-im
  579. Accept-Contact: *;language="en,fr"
  580. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  581. Organization: Doubango Telecom
  582.  
  583. <------------->
  584. --- (15 headers 0 lines) ---
  585. Scheduling destruction of SIP dialog '63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060' in 32000 ms (Method: BYE)
  586.  
  587. <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
  588. SIP/2.0 200 OK
  589. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport;received=192.168.88.187
  590. From: <sip:889@df7jal23ls0d.invalid>;tag=7NfJJeWFCMBeSuhOFJ3T
  591. To: "888"<sip:888@192.168.88.251>;tag=as176937a0
  592. Call-ID: 63d3c5027e7a3590019a503f1acecc48@192.168.88.251:5060
  593. CSeq: 43548 BYE
  594. Server: Asterisk PBX 13.2.0
  595. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  596. Supported: replaces, timer
  597. Content-Length: 0
  598.  
  599.  
  600. <------------>
  601.     -- Channel SIP/889-00000060 left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
  602.     -- Channel SIP/888-0000005f left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
  603.   == Spawn extension (default, 889, 1) exited non-zero on 'SIP/888-0000005f'
  604. Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)
  605. set_destination: Parsing <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws> for address/port to send to
  606. set_destination: URI is for WebSocket, we can't set destination
  607. Reliably Transmitting (no NAT) to 192.168.88.174:5060:
  608. BYE sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
  609. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
  610. Max-Forwards: 70
  611. From: <sip:889@192.168.88.251>;tag=as4947f672
  612. To: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  613. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  614. CSeq: 102 BYE
  615. User-Agent: Asterisk PBX 13.2.0
  616. Proxy-Authorization: Digest username="888", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="5a02570a", response="b05e0a15f788390a4b32dd0b4e0b3ec7"
  617. X-Asterisk-HangupCause: Normal Clearing
  618. X-Asterisk-HangupCauseCode: 16
  619. Content-Length: 0
  620.  
  621.  
  622. ---
  623.  
  624. <--- SIP read from WS:192.168.88.174:49537 --->
  625. SIP/2.0 200 OK
  626. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
  627. From: <sip:889@192.168.88.251>;tag=as4947f672
  628. To: "888"<sip:888@192.168.88.251>;tag=XeSJ7tTx1obTBnxuWVxo
  629. Contact: <sip:888@df7jal23ls0d.invalid;transport=ws>
  630. Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
  631. CSeq: 102 BYE
  632. Content-Length: 0
  633.  
  634. <------------->
  635. --- (8 headers 0 lines) ---
  636. SIP Response message for INCOMING dialog BYE arrived
  637. Really destroying SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' Method: INVITE
  638. Really destroying SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' Method: REGISTER
  639. Really destroying SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' Method: REGISTER
RAW Paste Data
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