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  1. SIP Debugging enabled
  2. Really destroying SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' Method: REGISTER
  3.  
  4. <--- SIP read from WS:192.168.88.174:49534 --->
  5. INVITE sip:889@192.168.88.251 SIP/2.0
  6. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport
  7. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  8. To: <sip:889@192.168.88.251>
  9. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
  10. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  11. CSeq: 50788 INVITE
  12. Content-Type: application/sdp
  13. Content-Length: 1591
  14. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  15. Max-Forwards: 70
  16. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  17. Organization: Doubango Telecom
  18.  
  19. v=0
  20. o=- 6673466038043864000 2 IN IP4 127.0.0.1
  21. s=Doubango Telecom - chrome
  22. t=0 0
  23. a=group:BUNDLE audio
  24. a=msid-semantic: WMS 1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
  25. m=audio 62984 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  26. c=IN IP4 192.168.88.174
  27. a=rtcp:62984 IN IP4 192.168.88.174
  28. a=candidate:159100432 1 udp 2122194687 192.168.88.174 62984 typ host generation 0
  29. a=candidate:159100432 2 udp 2122194687 192.168.88.174 62984 typ host generation 0
  30. a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  31. a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  32. a=ice-ufrag:508JZ2hjbUjZ3Hv3
  33. a=ice-pwd:pX+OETY08O8y4tswCjrFROTO
  34. a=ice-options:google-ice
  35. 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
  36. a=setup:actpass
  37. a=mid:audio
  38. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  39. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  40. a=sendrecv
  41. a=rtcp-mux
  42. a=rtpmap:111 opus/48000/2
  43. a=fmtp:111 minptime=10
  44. a=rtpmap:103 ISAC/16000
  45. a=rtpmap:104 ISAC/32000
  46. a=rtpmap:9 G722/8000
  47. a=rtpmap:0 PCMU/8000
  48. a=rtpmap:8 PCMA/8000
  49. a=rtpmap:106 CN/32000
  50. a=rtpmap:105 CN/16000
  51. a=rtpmap:13 CN/8000
  52. a=rtpmap:126 telephone-event/8000
  53. a=maxptime:60
  54. a=ssrc:4287587852 cname:F+ii1f0GVr8omEdf
  55. a=ssrc:4287587852 msid:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz 10bec123-e65f-45ed-a025-120b4b7859d4
  56. a=ssrc:4287587852 mslabel:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
  57. a=ssrc:4287587852 label:10bec123-e65f-45ed-a025-120b4b7859d4
  58. <------------->
  59. --- (13 headers 39 lines) ---
  60. Using INVITE request as basis request - 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  61. Found peer '888' for '888' from 192.168.88.174:49534
  62.  
  63. <--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
  64. SIP/2.0 401 Unauthorized
  65. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport;received=192.168.88.174
  66. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  67. To: <sip:889@192.168.88.251>;tag=as45cee7bb
  68. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  69. CSeq: 50788 INVITE
  70. Server: Asterisk PBX 13.2.0
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  72. Supported: replaces, timer
  73. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="3797a877"
  74. Content-Length: 0
  75.  
  76.  
  77. <------------>
  78. Scheduling destruction of SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' in 32000 ms (Method: INVITE)
  79.  
  80. <--- SIP read from WS:192.168.88.174:49534 --->
  81. ACK sip:889@192.168.88.251 SIP/2.0
  82. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport
  83. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  84. To: <sip:889@192.168.88.251>;tag=as45cee7bb
  85. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  86. CSeq: 50788 ACK
  87. Content-Length: 0
  88. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  89. Max-Forwards: 70
  90.  
  91. <------------->
  92. --- (9 headers 0 lines) ---
  93.  
  94. <--- SIP read from WS:192.168.88.174:49534 --->
  95. INVITE sip:889@192.168.88.251 SIP/2.0
  96. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport
  97. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  98. To: <sip:889@192.168.88.251>
  99. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
  100. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  101. CSeq: 50789 INVITE
  102. Content-Type: application/sdp
  103. Content-Length: 1591
  104. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  105. Max-Forwards: 70
  106. Authorization: Digest username="888",realm="192.168.88.251",nonce="3797a877",uri="sip:889@192.168.88.251",response="a7c984770475702277553f1cf5ffcda2",algorithm=MD5
  107. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  108. Organization: Doubango Telecom
  109.  
  110. v=0
  111. o=- 6673466038043864000 2 IN IP4 127.0.0.1
  112. s=Doubango Telecom - chrome
  113. t=0 0
  114. a=group:BUNDLE audio
  115. a=msid-semantic: WMS 1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
  116. m=audio 62984 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  117. c=IN IP4 192.168.88.174
  118. a=rtcp:62984 IN IP4 192.168.88.174
  119. a=candidate:159100432 1 udp 2122194687 192.168.88.174 62984 typ host generation 0
  120. a=candidate:159100432 2 udp 2122194687 192.168.88.174 62984 typ host generation 0
  121. a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  122. a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
  123. a=ice-ufrag:508JZ2hjbUjZ3Hv3
  124. a=ice-pwd:pX+OETY08O8y4tswCjrFROTO
  125. a=ice-options:google-ice
  126. 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
  127. a=setup:actpass
  128. a=mid:audio
  129. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  130. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  131. a=sendrecv
  132. a=rtcp-mux
  133. a=rtpmap:111 opus/48000/2
  134. a=fmtp:111 minptime=10
  135. a=rtpmap:103 ISAC/16000
  136. a=rtpmap:104 ISAC/32000
  137. a=rtpmap:9 G722/8000
  138. a=rtpmap:0 PCMU/8000
  139. a=rtpmap:8 PCMA/8000
  140. a=rtpmap:106 CN/32000
  141. a=rtpmap:105 CN/16000
  142. a=rtpmap:13 CN/8000
  143. a=rtpmap:126 telephone-event/8000
  144. a=maxptime:60
  145. a=ssrc:4287587852 cname:F+ii1f0GVr8omEdf
  146. a=ssrc:4287587852 msid:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz 10bec123-e65f-45ed-a025-120b4b7859d4
  147. a=ssrc:4287587852 mslabel:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
  148. a=ssrc:4287587852 label:10bec123-e65f-45ed-a025-120b4b7859d4
  149. <------------->
  150. --- (14 headers 39 lines) ---
  151. Using INVITE request as basis request - 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  152. Found peer '888' for '888' from 192.168.88.174:49534
  153. == Using SIP RTP CoS mark 5
  154. Found RTP audio format 111
  155. Found RTP audio format 103
  156. Found RTP audio format 104
  157. Found RTP audio format 9
  158. Found RTP audio format 0
  159. Found RTP audio format 8
  160. Found RTP audio format 106
  161. Found RTP audio format 105
  162. Found RTP audio format 13
  163. Found RTP audio format 126
  164. Found audio description format opus for ID 111
  165. Found unknown media description format ISAC for ID 103
  166. Found unknown media description format ISAC for ID 104
  167. Found audio description format G722 for ID 9
  168. Found audio description format PCMU for ID 0
  169. Found audio description format PCMA for ID 8
  170. Found unknown media description format CN for ID 106
  171. Found unknown media description format CN for ID 105
  172. Found audio description format CN for ID 13
  173. Found audio description format telephone-event for ID 126
  174. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  175. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  176. Peer audio RTP is at port 192.168.88.174:62984
  177. Looking for 889 in default (domain 192.168.88.251)
  178. sip_route_dump: route/path hop: <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>
  179.  
  180. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  181. SIP/2.0 100 Trying
  182. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
  183. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  184. To: <sip:889@192.168.88.251>
  185. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  186. CSeq: 50789 INVITE
  187. Server: Asterisk PBX 13.2.0
  188. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  189. Supported: replaces, timer
  190. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  191. Content-Length: 0
  192.  
  193.  
  194. <------------>
  195. -- Executing [889@default:1] Dial("SIP/888-00000057", "SIP/889") in new stack
  196. == Using SIP RTP CoS mark 5
  197. Audio is at 19510
  198. Adding codec ulaw to SDP
  199. Adding codec alaw to SDP
  200. Adding codec gsm to SDP
  201. Adding non-codec 0x1 (telephone-event) to SDP
  202. Reliably Transmitting (no NAT) to 192.168.88.187:49625:
  203. INVITE sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
  204. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
  205. Max-Forwards: 70
  206. From: "888" <sip:888@192.168.88.251>;tag=as31b619ed
  207. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  208. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  209. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  210. CSeq: 102 INVITE
  211. User-Agent: Asterisk PBX 13.2.0
  212. Date: Fri, 13 Feb 2015 04:08:06 GMT
  213. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  214. Supported: replaces, timer
  215. Content-Type: application/sdp
  216. Content-Length: 674
  217.  
  218. v=0
  219. o=root 258976009 258976009 IN IP4 192.168.88.251
  220. s=Asterisk PBX 13.2.0
  221. c=IN IP4 192.168.88.251
  222. t=0 0
  223. m=audio 19510 RTP/SAVPF 0 8 3 101
  224. a=rtpmap:0 PCMU/8000
  225. a=rtpmap:8 PCMA/8000
  226. a=rtpmap:3 GSM/8000
  227. a=rtpmap:101 telephone-event/8000
  228. a=fmtp:101 0-16
  229. a=maxptime:150
  230. a=ice-ufrag:55783ab604a96c221e1f9fca4b526dd4
  231. a=ice-pwd:68e6006a5653d426317ee5fe2172d4c9
  232. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 19510 typ host
  233. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 19511 typ host
  234. a=connection:new
  235. a=setup:actpass
  236. 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
  237. a=sendrecv
  238.  
  239. ---
  240. -- Called SIP/889
  241.  
  242. <--- SIP read from WS:192.168.88.187:49625 --->
  243. SIP/2.0 100 Trying (sent from the Transaction Layer)
  244. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
  245. From: "888"<sip:888@192.168.88.251>;tag=as31b619ed
  246. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
  247. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  248. CSeq: 102 INVITE
  249. Content-Length: 0
  250.  
  251. <------------->
  252. --- (7 headers 0 lines) ---
  253.  
  254. <--- SIP read from WS:192.168.88.187:49625 --->
  255. SIP/2.0 180 Ringing
  256. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
  257. From: "888"<sip:888@192.168.88.251>;tag=as31b619ed
  258. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
  259. Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
  260. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  261. CSeq: 102 INVITE
  262. Content-Length: 0
  263. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  264.  
  265. <------------->
  266. --- (9 headers 0 lines) ---
  267. sip_route_dump: route/path hop: <sip:889@df7jal23ls0d.invalid;transport=ws>
  268. -- SIP/889-00000058 is ringing
  269.  
  270. <--- Transmitting (no NAT) to 192.168.88.174:5060 --->
  271. SIP/2.0 180 Ringing
  272. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
  273. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  274. To: <sip:889@192.168.88.251>;tag=as0c09af77
  275. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  276. CSeq: 50789 INVITE
  277. Server: Asterisk PBX 13.2.0
  278. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  279. Supported: replaces, timer
  280. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  281. Content-Length: 0
  282.  
  283.  
  284. <------------>
  285. Really destroying SIP dialog '01e934ff-5052-fc68-6a3c-0bc36705ff46' Method: REGISTER
  286.  
  287. <--- SIP read from WS:192.168.88.187:49625 --->
  288. SIP/2.0 200 OK
  289. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
  290. From: "888"<sip:888@192.168.88.251>;tag=as31b619ed
  291. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
  292. Contact: <sip:889@df7jal23ls0d.invalid;transport=ws>
  293. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  294. CSeq: 102 INVITE
  295. Content-Type: application/sdp
  296. Content-Length: 1167
  297. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  298.  
  299. v=0
  300. o=- 5364208010358385000 2 IN IP4 127.0.0.1
  301. s=Doubango Telecom - chrome
  302. t=0 0
  303. a=msid-semantic: WMS r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G
  304. m=audio 50364 UDP/TLS/RTP/SAVPF 0 8 101
  305. c=IN IP4 192.168.88.187
  306. a=rtcp:50365 IN IP4 192.168.88.187
  307. a=candidate:2577307183 1 udp 2122194687 192.168.88.187 50364 typ host generation 0
  308. a=candidate:2577307183 2 udp 2122194686 192.168.88.187 50365 typ host generation 0
  309. a=candidate:3609029343 1 tcp 1518214911 192.168.88.187 0 typ host tcptype active generation 0
  310. a=candidate:3609029343 2 tcp 1518214910 192.168.88.187 0 typ host tcptype active generation 0
  311. a=ice-ufrag:GefBqC5O5qnV6nwt
  312. a=ice-pwd:QzoXPid0PCMwYEtzJ2rZpbHJ
  313. 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
  314. a=setup:active
  315. a=mid:audio
  316. a=sendrecv
  317. a=rtpmap:0 PCMU/8000
  318. a=rtpmap:8 PCMA/8000
  319. a=rtpmap:101 telephone-event/8000
  320. a=ssrc:99136968 cname:Cr5VtuG5xK5gmOI0
  321. a=ssrc:99136968 msid:r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G 0376de4e-6e18-4ca8-b7e7-f479bbd58814
  322. a=ssrc:99136968 mslabel:r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G
  323. a=ssrc:99136968 label:0376de4e-6e18-4ca8-b7e7-f479bbd58814
  324. <------------->
  325. --- (10 headers 25 lines) ---
  326. Found RTP audio format 0
  327. Found RTP audio format 8
  328. Found RTP audio format 101
  329. Found audio description format PCMU for ID 0
  330. Found audio description format PCMA for ID 8
  331. Found audio description format telephone-event for ID 101
  332. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  333. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  334. Peer audio RTP is at port 192.168.88.187:50364
  335. sip_route_dump: route/path hop: <sip:889@df7jal23ls0d.invalid;transport=ws>
  336. [Feb 13 06:08:10] ERROR[1055][C-0000002d]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  337. [Feb 13 06:08:10] WARNING[1055][C-0000002d]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  338. set_destination: Parsing <sip:889@df7jal23ls0d.invalid;transport=ws> for address/port to send to
  339. set_destination: URI is for WebSocket, we can't set destination
  340. Transmitting (no NAT) to 192.168.88.187:49625:
  341. ACK sip:889@df7jal23ls0d.invalid;transport=ws SIP/2.0
  342. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK00e39beb
  343. Max-Forwards: 70
  344. From: "888" <sip:888@192.168.88.251>;tag=as31b619ed
  345. To: <sip:889@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
  346. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  347. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  348. CSeq: 102 ACK
  349. User-Agent: Asterisk PBX 13.2.0
  350. Content-Length: 0
  351.  
  352.  
  353. ---
  354. -- SIP/889-00000058 answered SIP/888-00000057
  355. Audio is at 18654
  356. Adding codec ulaw to SDP
  357. Adding codec alaw to SDP
  358. Adding codec gsm to SDP
  359. Adding non-codec 0x1 (telephone-event) to SDP
  360.  
  361. <--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
  362. SIP/2.0 200 OK
  363. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
  364. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  365. To: <sip:889@192.168.88.251>;tag=as0c09af77
  366. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  367. CSeq: 50789 INVITE
  368. Server: Asterisk PBX 13.2.0
  369. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  370. Supported: replaces, timer
  371. Contact: <sip:889@192.168.88.251:5060;transport=WS>
  372. Content-Type: application/sdp
  373. Content-Length: 673
  374.  
  375. v=0
  376. o=root 645993891 645993891 IN IP4 192.168.88.251
  377. s=Asterisk PBX 13.2.0
  378. c=IN IP4 192.168.88.251
  379. t=0 0
  380. m=audio 18654 RTP/SAVPF 0 8 3 126
  381. a=rtpmap:0 PCMU/8000
  382. a=rtpmap:8 PCMA/8000
  383. a=rtpmap:3 GSM/8000
  384. a=rtpmap:126 telephone-event/8000
  385. a=fmtp:126 0-16
  386. a=maxptime:150
  387. a=ice-ufrag:66a8fa20406da1b05cc9891248f19665
  388. a=ice-pwd:4f5fa9126a825b6f38f1b4d515068a62
  389. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 18654 typ host
  390. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 18655 typ host
  391. a=connection:new
  392. a=setup:active
  393. 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
  394. a=sendrecv
  395.  
  396. <------------>
  397. -- Channel SIP/888-00000057 joined 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
  398. -- Channel SIP/889-00000058 joined 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
  399.  
  400. <--- SIP read from WS:192.168.88.174:49534 --->
  401. ACK sip:889@192.168.88.251:5060;transport=WS SIP/2.0
  402. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxlSp7r6AGZ9XNFnEio78;rport
  403. From: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  404. To: <sip:889@192.168.88.251>;tag=as0c09af77
  405. Contact: "888"<sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  406. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  407. CSeq: 50789 ACK
  408. Content-Length: 0
  409. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  410. Max-Forwards: 70
  411. Authorization: Digest username="888",realm="192.168.88.251",nonce="3797a877",uri="sip:889@192.168.88.251:5060;transport=WS",response="9ffc84a96b9cdabc77990888b9fa8ab6",algorithm=MD5
  412. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  413. Organization: Doubango Telecom
  414.  
  415. <------------->
  416. --- (13 headers 0 lines) ---
  417. > 0x7fd8389ed7e0 -- Probation passed - setting RTP source address to 192.168.88.187:50364
  418. > 0x7fd8388a9ba0 -- Probation passed - setting RTP source address to 192.168.88.174:62984
  419.  
  420. <--- SIP read from WS:192.168.88.187:49625 --->
  421. BYE sip:888@192.168.88.251:5060;transport=WS SIP/2.0
  422. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNMvLQe0eTKLVlTNvykfChDzK6nPugf9n;rport
  423. From: <sip:889@df7jal23ls0d.invalid>;tag=gwK0qRuZMIfigmgTWstQ
  424. To: "888"<sip:888@192.168.88.251>;tag=as31b619ed
  425. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  426. CSeq: 58758 BYE
  427. Content-Length: 0
  428. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  429. Max-Forwards: 70
  430. Accept-Contact: *;+g.oma.sip-im
  431. Accept-Contact: *;language="en,fr"
  432. Accept-Contact: *;+g.oma.sip-im
  433. Accept-Contact: *;language="en,fr"
  434. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  435. Organization: Doubango Telecom
  436.  
  437. <------------->
  438. --- (15 headers 0 lines) ---
  439. Scheduling destruction of SIP dialog '4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060' in 32000 ms (Method: BYE)
  440.  
  441. <--- Transmitting (no NAT) to 192.168.88.187:5060 --->
  442. SIP/2.0 200 OK
  443. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNMvLQe0eTKLVlTNvykfChDzK6nPugf9n;rport;received=192.168.88.187
  444. From: <sip:889@df7jal23ls0d.invalid>;tag=gwK0qRuZMIfigmgTWstQ
  445. To: "888"<sip:888@192.168.88.251>;tag=as31b619ed
  446. Call-ID: 4e3ca96a1f11809a0c12f7983b12eef7@192.168.88.251:5060
  447. CSeq: 58758 BYE
  448. Server: Asterisk PBX 13.2.0
  449. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  450. Supported: replaces, timer
  451. Content-Length: 0
  452.  
  453.  
  454. <------------>
  455. -- Channel SIP/889-00000058 left 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
  456. -- Channel SIP/888-00000057 left 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
  457. == Spawn extension (default, 889, 1) exited non-zero on 'SIP/888-00000057'
  458. Scheduling destruction of SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' in 32000 ms (Method: INVITE)
  459. set_destination: Parsing <sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws> for address/port to send to
  460. set_destination: URI is for WebSocket, we can't set destination
  461. Reliably Transmitting (no NAT) to 192.168.88.174:5060:
  462. BYE sip:888@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
  463. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK5e8b871a
  464. Max-Forwards: 70
  465. From: <sip:889@192.168.88.251>;tag=as0c09af77
  466. To: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  467. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  468. CSeq: 102 BYE
  469. User-Agent: Asterisk PBX 13.2.0
  470. Proxy-Authorization: Digest username="888", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="3797a877", response="a04d3572e794d02d67f953d6bea7311d"
  471. X-Asterisk-HangupCause: Normal Clearing
  472. X-Asterisk-HangupCauseCode: 16
  473. Content-Length: 0
  474.  
  475.  
  476. ---
  477.  
  478. <--- SIP read from WS:192.168.88.174:49534 --->
  479. SIP/2.0 200 OK
  480. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK5e8b871a
  481. From: <sip:889@192.168.88.251>;tag=as0c09af77
  482. To: "888"<sip:888@192.168.88.251>;tag=Jl1s49meNfTyNH6uJ7AI
  483. Contact: <sip:888@df7jal23ls0d.invalid;transport=ws>
  484. Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
  485. CSeq: 102 BYE
  486. Content-Length: 0
  487.  
  488. <------------->
  489. --- (8 headers 0 lines) ---
  490. SIP Response message for INCOMING dialog BYE arrived
  491. Really destroying SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' Method: INVITE
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