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  1. == Using SIP RTP CoS mark 5
  2. -- Executing [893@default:1] Dial("SIP/892-0000009e", "SIP/893") in new stack
  3. == Using SIP RTP CoS mark 5
  4. -- Called SIP/893
  5. -- SIP/893-0000009f is ringing
  6. [Feb 18 04:05:19] ERROR[5857][C-0000006d]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  7. [Feb 18 04:05:19] WARNING[5857][C-0000006d]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  8. -- SIP/893-0000009f answered SIP/892-0000009e
  9. -- Channel SIP/892-0000009e joined 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
  10. -- Channel SIP/893-0000009f joined 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
  11. > 0x7fe19d3a6080 -- Probation passed - setting RTP source address to 192.168.88.189:51710
  12. > 0x7fe19c49f1c0 -- Probation passed - setting RTP source address to 192.168.88.188:60363
  13. -- Channel SIP/893-0000009f left 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
  14. -- Channel SIP/892-0000009e left 'simple_bridge' basic-bridge <e5ba6c48-c243-4a1b-976a-6b26fad644db>
  15. == Spawn extension (default, 893, 1) exited non-zero on 'SIP/892-0000009e'
  16. ast*CLI> sip set debug on
  17. SIP Debugging enabled
  18.  
  19. <--- SIP read from WS:192.168.88.188:49766 --->
  20. INVITE sip:[email protected] SIP/2.0
  21. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport
  22. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  23. To: <sip:[email protected]>
  24. Contact: "892"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  25. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  26. CSeq: 65211 INVITE
  27. Content-Type: application/sdp
  28. Content-Length: 1593
  29. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  30. Max-Forwards: 70
  31. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  32. Organization: Doubango Telecom
  33.  
  34. v=0
  35. o=- 9013918505325537000 2 IN IP4 127.0.0.1
  36. s=Doubango Telecom - chrome
  37. t=0 0
  38. a=group:BUNDLE audio
  39. a=msid-semantic: WMS hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
  40. m=audio 60364 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  41. c=IN IP4 192.168.88.188
  42. a=rtcp:60364 IN IP4 192.168.88.188
  43. a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60364 typ host generation 0
  44. a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60364 typ host generation 0
  45. a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  46. a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  47. a=ice-ufrag:8mZ6nJPeh4HNMJ8u
  48. a=ice-pwd:RGdhONKCR6yEsEs8H/hiQx0U
  49. a=ice-options:google-ice
  50. a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
  51. a=setup:actpass
  52. a=mid:audio
  53. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  54. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  55. a=sendrecv
  56. a=rtcp-mux
  57. a=rtpmap:111 opus/48000/2
  58. a=fmtp:111 minptime=10
  59. a=rtpmap:103 ISAC/16000
  60. a=rtpmap:104 ISAC/32000
  61. a=rtpmap:9 G722/8000
  62. a=rtpmap:0 PCMU/8000
  63. a=rtpmap:8 PCMA/8000
  64. a=rtpmap:106 CN/32000
  65. a=rtpmap:105 CN/16000
  66. a=rtpmap:13 CN/8000
  67. a=rtpmap:126 telephone-event/8000
  68. a=maxptime:60
  69. a=ssrc:2479575733 cname:URhPGkQlkeqtQHWw
  70. a=ssrc:2479575733 msid:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE 64ee14c7-d87c-431d-8ed4-713c3117a869
  71. a=ssrc:2479575733 mslabel:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
  72. a=ssrc:2479575733 label:64ee14c7-d87c-431d-8ed4-713c3117a869
  73. <------------->
  74. --- (13 headers 39 lines) ---
  75. Using INVITE request as basis request - 7252dd20-3201-26b9-e94f-1754d3b61c84
  76. Found peer '892' for '892' from 192.168.88.188:49766
  77.  
  78. <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
  79. SIP/2.0 401 Unauthorized
  80. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport;received=192.168.88.188
  81. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  82. To: <sip:[email protected]>;tag=as43bcc41a
  83. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  84. CSeq: 65211 INVITE
  85. Server: Asterisk PBX 13.2.0
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  87. Supported: replaces, timer
  88. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="1cded12e"
  89. Content-Length: 0
  90.  
  91.  
  92. <------------>
  93. Scheduling destruction of SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' in 32000 ms (Method: INVITE)
  94.  
  95. <--- SIP read from WS:192.168.88.188:49766 --->
  96. ACK sip:[email protected] SIP/2.0
  97. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTdphAFzlyRsZMc5LmuSIzPtCPTl1zFc;rport
  98. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  99. To: <sip:[email protected]>;tag=as43bcc41a
  100. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  101. CSeq: 65211 ACK
  102. Content-Length: 0
  103. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  104. Max-Forwards: 70
  105.  
  106. <------------->
  107. --- (9 headers 0 lines) ---
  108.  
  109. <--- SIP read from WS:192.168.88.188:49766 --->
  110. INVITE sip:[email protected] SIP/2.0
  111. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport
  112. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  113. To: <sip:[email protected]>
  114. Contact: "892"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  115. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  116. CSeq: 65212 INVITE
  117. Content-Type: application/sdp
  118. Content-Length: 1593
  119. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  120. Max-Forwards: 70
  121. Authorization: Digest username="892",realm="192.168.88.251",nonce="1cded12e",uri="sip:[email protected]",response="47dd36205a3c74b2fd4b21a1521dac3f",algorithm=MD5
  122. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  123. Organization: Doubango Telecom
  124.  
  125. v=0
  126. o=- 9013918505325537000 2 IN IP4 127.0.0.1
  127. s=Doubango Telecom - chrome
  128. t=0 0
  129. a=group:BUNDLE audio
  130. a=msid-semantic: WMS hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
  131. m=audio 60364 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  132. c=IN IP4 192.168.88.188
  133. a=rtcp:60364 IN IP4 192.168.88.188
  134. a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60364 typ host generation 0
  135. a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60364 typ host generation 0
  136. a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  137. a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  138. a=ice-ufrag:8mZ6nJPeh4HNMJ8u
  139. a=ice-pwd:RGdhONKCR6yEsEs8H/hiQx0U
  140. a=ice-options:google-ice
  141. a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
  142. a=setup:actpass
  143. a=mid:audio
  144. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  145. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  146. a=sendrecv
  147. a=rtcp-mux
  148. a=rtpmap:111 opus/48000/2
  149. a=fmtp:111 minptime=10
  150. a=rtpmap:103 ISAC/16000
  151. a=rtpmap:104 ISAC/32000
  152. a=rtpmap:9 G722/8000
  153. a=rtpmap:0 PCMU/8000
  154. a=rtpmap:8 PCMA/8000
  155. a=rtpmap:106 CN/32000
  156. a=rtpmap:105 CN/16000
  157. a=rtpmap:13 CN/8000
  158. a=rtpmap:126 telephone-event/8000
  159. a=maxptime:60
  160. a=ssrc:2479575733 cname:URhPGkQlkeqtQHWw
  161. a=ssrc:2479575733 msid:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE 64ee14c7-d87c-431d-8ed4-713c3117a869
  162. a=ssrc:2479575733 mslabel:hEEmP96XY2MOoU8tZSf60S91D38RCTT6JUmE
  163. a=ssrc:2479575733 label:64ee14c7-d87c-431d-8ed4-713c3117a869
  164. <------------->
  165. --- (14 headers 39 lines) ---
  166. Using INVITE request as basis request - 7252dd20-3201-26b9-e94f-1754d3b61c84
  167. Found peer '892' for '892' from 192.168.88.188:49766
  168. == Using SIP RTP CoS mark 5
  169. Found RTP audio format 111
  170. Found RTP audio format 103
  171. Found RTP audio format 104
  172. Found RTP audio format 9
  173. Found RTP audio format 0
  174. Found RTP audio format 8
  175. Found RTP audio format 106
  176. Found RTP audio format 105
  177. Found RTP audio format 13
  178. Found RTP audio format 126
  179. Found audio description format opus for ID 111
  180. Found unknown media description format ISAC for ID 103
  181. Found unknown media description format ISAC for ID 104
  182. Found audio description format G722 for ID 9
  183. Found audio description format PCMU for ID 0
  184. Found audio description format PCMA for ID 8
  185. Found unknown media description format CN for ID 106
  186. Found unknown media description format CN for ID 105
  187. Found audio description format CN for ID 13
  188. Found audio description format telephone-event for ID 126
  189. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  190. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  191. Peer audio RTP is at port 192.168.88.188:60364
  192. Looking for 893 in default (domain 192.168.88.251)
  193. sip_route_dump: route/path hop: <sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>
  194.  
  195. <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
  196. SIP/2.0 100 Trying
  197. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
  198. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  199. To: <sip:[email protected]>
  200. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  201. CSeq: 65212 INVITE
  202. Server: Asterisk PBX 13.2.0
  203. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  204. Supported: replaces, timer
  205. Contact: <sip:[email protected]:5060;transport=WS>
  206. Content-Length: 0
  207.  
  208.  
  209. <------------>
  210. -- Executing [893@default:1] Dial("SIP/892-000000a0", "SIP/893") in new stack
  211. == Using SIP RTP CoS mark 5
  212. Audio is at 16278
  213. Adding codec ulaw to SDP
  214. Adding codec alaw to SDP
  215. Adding codec gsm to SDP
  216. Adding non-codec 0x1 (telephone-event) to SDP
  217. Reliably Transmitting (no NAT) to 192.168.88.189:54237:
  218. INVITE sip:[email protected];rtcweb-breaker=no;transport=ws SIP/2.0
  219. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
  220. Max-Forwards: 70
  221. From: "892" <sip:[email protected]>;tag=as3d7652ce
  222. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>
  223. Contact: <sip:[email protected]:5060;transport=WS>
  224. Call-ID: [email protected]:5060
  225. CSeq: 102 INVITE
  226. User-Agent: Asterisk PBX 13.2.0
  227. Date: Wed, 18 Feb 2015 02:05:35 GMT
  228. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  229. Supported: replaces, timer
  230. Content-Type: application/sdp
  231. Content-Length: 674
  232.  
  233. v=0
  234. o=root 250873218 250873218 IN IP4 192.168.88.251
  235. s=Asterisk PBX 13.2.0
  236. c=IN IP4 192.168.88.251
  237. t=0 0
  238. m=audio 16278 RTP/SAVPF 0 8 3 101
  239. a=rtpmap:0 PCMU/8000
  240. a=rtpmap:8 PCMA/8000
  241. a=rtpmap:3 GSM/8000
  242. a=rtpmap:101 telephone-event/8000
  243. a=fmtp:101 0-16
  244. a=maxptime:150
  245. a=ice-ufrag:7ae4574e11e0910e742f5f4c4503d85e
  246. a=ice-pwd:0ba5d6ad73334552538a52503bc1da69
  247. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 16278 typ host
  248. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 16279 typ host
  249. a=connection:new
  250. a=setup:actpass
  251. 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
  252. a=sendrecv
  253.  
  254. ---
  255. -- Called SIP/893
  256.  
  257. <--- SIP read from WS:192.168.88.189:54237 --->
  258. SIP/2.0 100 Trying (sent from the Transaction Layer)
  259. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
  260. From: "892"<sip:[email protected]>;tag=as3d7652ce
  261. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>
  262. Call-ID: [email protected]:5060
  263. CSeq: 102 INVITE
  264. Content-Length: 0
  265.  
  266. <------------->
  267. --- (7 headers 0 lines) ---
  268.  
  269. <--- SIP read from WS:192.168.88.189:54237 --->
  270. SIP/2.0 180 Ringing
  271. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
  272. From: "892"<sip:[email protected]>;tag=as3d7652ce
  273. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
  274. Contact: <sip:[email protected];transport=ws>
  275. Call-ID: [email protected]:5060
  276. CSeq: 102 INVITE
  277. Content-Length: 0
  278. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  279.  
  280. <------------->
  281. --- (9 headers 0 lines) ---
  282. sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
  283. -- SIP/893-000000a1 is ringing
  284.  
  285. <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
  286. SIP/2.0 180 Ringing
  287. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
  288. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  289. To: <sip:[email protected]>;tag=as1a240a25
  290. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  291. CSeq: 65212 INVITE
  292. Server: Asterisk PBX 13.2.0
  293. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  294. Supported: replaces, timer
  295. Contact: <sip:[email protected]:5060;transport=WS>
  296. Content-Length: 0
  297.  
  298.  
  299. <------------>
  300.  
  301. <--- SIP read from WS:192.168.88.189:54237 --->
  302. REGISTER sip:192.168.88.251 SIP/2.0
  303. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhXsTmRcEND87oBsyUPTp9l2qrDJsKw2x;rport
  304. From: "893"<sip:[email protected]>;tag=LH6X6SThWalMGcJtquUq
  305. To: "893"<sip:[email protected]>
  306. Contact: "893"<sip:[email protected];rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  307. Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
  308. CSeq: 27463 REGISTER
  309. Content-Length: 0
  310. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  311. Max-Forwards: 70
  312. Authorization: Digest username="893",realm="192.168.88.251",nonce="3862f2d8",uri="sip:192.168.88.251",response="87ce2c4b8f87482ab0754ece3dd40052",algorithm=MD5
  313. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  314. Organization: Doubango Telecom
  315.  
  316. <------------->
  317. --- (13 headers 0 lines) ---
  318.  
  319. <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
  320. SIP/2.0 401 Unauthorized
  321. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhXsTmRcEND87oBsyUPTp9l2qrDJsKw2x;rport;received=192.168.88.189
  322. From: "893"<sip:[email protected]>;tag=LH6X6SThWalMGcJtquUq
  323. To: "893"<sip:[email protected]>;tag=as2608aa5c
  324. Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
  325. CSeq: 27463 REGISTER
  326. Server: Asterisk PBX 13.2.0
  327. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  328. Supported: replaces, timer
  329. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="1d777b46"
  330. Content-Length: 0
  331.  
  332.  
  333. <------------>
  334. Scheduling destruction of SIP dialog '5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc' in 32000 ms (Method: REGISTER)
  335.  
  336. <--- SIP read from WS:192.168.88.189:54237 --->
  337. REGISTER sip:192.168.88.251 SIP/2.0
  338. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOCZLWAod2T2lWC6yOUiCFTK3uAAQQvsd;rport
  339. From: "893"<sip:[email protected]>;tag=LH6X6SThWalMGcJtquUq
  340. To: "893"<sip:[email protected]>
  341. Contact: "893"<sip:[email protected];rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
  342. Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
  343. CSeq: 27464 REGISTER
  344. Content-Length: 0
  345. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  346. Max-Forwards: 70
  347. Authorization: Digest username="893",realm="192.168.88.251",nonce="1d777b46",uri="sip:192.168.88.251",response="83f9e3fef5a8d6ca80eed28e40f86fca",algorithm=MD5
  348. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  349. Organization: Doubango Telecom
  350.  
  351. <------------->
  352. --- (13 headers 0 lines) ---
  353.  
  354. <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
  355. SIP/2.0 200 OK
  356. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOCZLWAod2T2lWC6yOUiCFTK3uAAQQvsd;rport;received=192.168.88.189
  357. From: "893"<sip:[email protected]>;tag=LH6X6SThWalMGcJtquUq
  358. To: "893"<sip:[email protected]>;tag=as2608aa5c
  359. Call-ID: 5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc
  360. CSeq: 27464 REGISTER
  361. Server: Asterisk PBX 13.2.0
  362. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  363. Supported: replaces, timer
  364. Expires: 200
  365. Contact: <sip:[email protected];rtcweb-breaker=no;transport=ws>;expires=200
  366. Date: Wed, 18 Feb 2015 02:05:35 GMT
  367. Content-Length: 0
  368.  
  369.  
  370. <------------>
  371. Scheduling destruction of SIP dialog '5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc' in 32000 ms (Method: REGISTER)
  372.  
  373. <--- SIP read from WS:192.168.88.189:54237 --->
  374. SIP/2.0 200 OK
  375. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK591d615f
  376. From: "892"<sip:[email protected]>;tag=as3d7652ce
  377. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
  378. Contact: <sip:[email protected];transport=ws>
  379. Call-ID: [email protected]:5060
  380. CSeq: 102 INVITE
  381. Content-Type: application/sdp
  382. Content-Length: 1173
  383. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  384.  
  385. v=0
  386. o=- 6851900569207422000 2 IN IP4 127.0.0.1
  387. s=Doubango Telecom - chrome
  388. t=0 0
  389. a=msid-semantic: WMS 5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa
  390. m=audio 51712 UDP/TLS/RTP/SAVPF 0 8 101
  391. c=IN IP4 192.168.88.189
  392. a=rtcp:51713 IN IP4 192.168.88.189
  393. a=candidate:2034360604 1 udp 2122194687 192.168.88.189 51712 typ host generation 0
  394. a=candidate:2034360604 2 udp 2122194686 192.168.88.189 51713 typ host generation 0
  395. a=candidate:935468524 1 tcp 1518214911 192.168.88.189 0 typ host tcptype active generation 0
  396. a=candidate:935468524 2 tcp 1518214910 192.168.88.189 0 typ host tcptype active generation 0
  397. a=ice-ufrag:17Hg6cuOE6d2IqDO
  398. a=ice-pwd:bguLRJ63C1EopK+2L82QoRMh
  399. a=fingerprint:sha-256 33:7A:5F:05:75:AB:62:A6:20:78:D5:F7:EF:BB:BB:3A:0A:D2:89:0F:DA:4D:22:50:8C:E7:90:A4:51:34:F8:61
  400. a=setup:active
  401. a=mid:audio
  402. a=sendrecv
  403. a=rtpmap:0 PCMU/8000
  404. a=rtpmap:8 PCMA/8000
  405. a=rtpmap:101 telephone-event/8000
  406. a=ssrc:3629716700 cname:YYEK0OxOe3UE1N+f
  407. a=ssrc:3629716700 msid:5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa 0187d8f9-44c3-4140-8802-7381e4e403e8
  408. a=ssrc:3629716700 mslabel:5eDeNziUzTyYfgV9E2NauG2fmuJIFHy7aFsa
  409. a=ssrc:3629716700 label:0187d8f9-44c3-4140-8802-7381e4e403e8
  410. <------------->
  411. --- (10 headers 25 lines) ---
  412. Found RTP audio format 0
  413. Found RTP audio format 8
  414. Found RTP audio format 101
  415. Found audio description format PCMU for ID 0
  416. Found audio description format PCMA for ID 8
  417. Found audio description format telephone-event for ID 101
  418. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  419. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  420. Peer audio RTP is at port 192.168.88.189:51712
  421. sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
  422. [Feb 18 04:05:39] ERROR[5857][C-0000006e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  423. [Feb 18 04:05:39] WARNING[5857][C-0000006e]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  424. set_destination: Parsing <sip:[email protected];transport=ws> for address/port to send to
  425. set_destination: URI is for WebSocket, we can't set destination
  426. Transmitting (no NAT) to 192.168.88.189:54237:
  427. ACK sip:[email protected];transport=ws SIP/2.0
  428. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3b200041
  429. Max-Forwards: 70
  430. From: "892" <sip:[email protected]>;tag=as3d7652ce
  431. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>;tag=i1J8u8fbSqqmTOnOGRim
  432. Contact: <sip:[email protected]:5060;transport=WS>
  433. Call-ID: [email protected]:5060
  434. CSeq: 102 ACK
  435. User-Agent: Asterisk PBX 13.2.0
  436. Content-Length: 0
  437.  
  438.  
  439. ---
  440. -- SIP/893-000000a1 answered SIP/892-000000a0
  441. Audio is at 13722
  442. Adding codec ulaw to SDP
  443. Adding codec alaw to SDP
  444. Adding codec gsm to SDP
  445. Adding non-codec 0x1 (telephone-event) to SDP
  446.  
  447. <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
  448. SIP/2.0 200 OK
  449. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKoZlAVVmrdRXkdWcQvb3KLhn9fe3shI5;rport;received=192.168.88.188
  450. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  451. To: <sip:[email protected]>;tag=as1a240a25
  452. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  453. CSeq: 65212 INVITE
  454. Server: Asterisk PBX 13.2.0
  455. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  456. Supported: replaces, timer
  457. Contact: <sip:[email protected]:5060;transport=WS>
  458. Content-Type: application/sdp
  459. Content-Length: 673
  460.  
  461. v=0
  462. o=root 382034728 382034728 IN IP4 192.168.88.251
  463. s=Asterisk PBX 13.2.0
  464. c=IN IP4 192.168.88.251
  465. t=0 0
  466. m=audio 13722 RTP/SAVPF 0 8 3 126
  467. a=rtpmap:0 PCMU/8000
  468. a=rtpmap:8 PCMA/8000
  469. a=rtpmap:3 GSM/8000
  470. a=rtpmap:126 telephone-event/8000
  471. a=fmtp:126 0-16
  472. a=maxptime:150
  473. a=ice-ufrag:5a9b3168230d32be01020e2b48e838b2
  474. a=ice-pwd:31ddd4366598156e7885fd1d1932870f
  475. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 13722 typ host
  476. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 13723 typ host
  477. a=connection:new
  478. a=setup:active
  479. 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
  480. a=sendrecv
  481.  
  482. <------------>
  483. -- Channel SIP/892-000000a0 joined 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
  484. -- Channel SIP/893-000000a1 joined 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
  485.  
  486. <--- SIP read from WS:192.168.88.188:49766 --->
  487. ACK sip:[email protected]:5060;transport=WS SIP/2.0
  488. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKl6PTQTDyGPWBoThqSJ15;rport
  489. From: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  490. To: <sip:[email protected]>;tag=as1a240a25
  491. Contact: "892"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  492. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  493. CSeq: 65212 ACK
  494. Content-Length: 0
  495. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  496. Max-Forwards: 70
  497. Authorization: Digest username="892",realm="192.168.88.251",nonce="1cded12e",uri="sip:[email protected]:5060;transport=WS",response="55ff9d4edd610e4c776a49ae4c3b79b0",algorithm=MD5
  498. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  499. Organization: Doubango Telecom
  500.  
  501. <------------->
  502. --- (13 headers 0 lines) ---
  503. > 0x7fe19d38c3e0 -- Probation passed - setting RTP source address to 192.168.88.189:51712
  504. > 0x7fe19c49f1c0 -- Probation passed - setting RTP source address to 192.168.88.188:60364
  505. Really destroying SIP dialog 'dc8a463e-01f5-927f-6d0f-31c24e5c091e' Method: REGISTER
  506.  
  507. <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
  508. SIP/2.0 180 Ringing
  509. Via: SIP/2.0/WS 192.0.2.157;branch=z9hG4bK4472428;received=192.168.88.182
  510. From: <sip:[email protected]>;tag=grberh41iu
  511. To: <sip:[email protected]>;tag=as35a3c936
  512. Call-ID: 61r71uq6tdlnd58mga7l
  513. CSeq: 5841 INVITE
  514. Server: Asterisk PBX 13.2.0
  515. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  516. Supported: replaces, timer
  517. Contact: <sip:[email protected]:5060;transport=WS>
  518. Content-Length: 0
  519.  
  520.  
  521. <------------>
  522.  
  523. <--- SIP read from WS:192.168.88.189:54237 --->
  524. BYE sip:[email protected]:5060;transport=WS SIP/2.0
  525. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxMk3DoB6czCTDkP8KmlAmfpIEEOCPnGp;rport
  526. From: <sip:[email protected]>;tag=i1J8u8fbSqqmTOnOGRim
  527. To: "892"<sip:[email protected]>;tag=as3d7652ce
  528. Call-ID: [email protected]:5060
  529. CSeq: 2151 BYE
  530. Content-Length: 0
  531. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  532. Max-Forwards: 70
  533. Accept-Contact: *;+g.oma.sip-im
  534. Accept-Contact: *;language="en,fr"
  535. Accept-Contact: *;+g.oma.sip-im
  536. Accept-Contact: *;language="en,fr"
  537. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  538. Organization: Doubango Telecom
  539.  
  540. <------------->
  541. --- (15 headers 0 lines) ---
  542. Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)
  543.  
  544. <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
  545. SIP/2.0 200 OK
  546. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxMk3DoB6czCTDkP8KmlAmfpIEEOCPnGp;rport;received=192.168.88.189
  547. From: <sip:[email protected]>;tag=i1J8u8fbSqqmTOnOGRim
  548. To: "892"<sip:[email protected]>;tag=as3d7652ce
  549. Call-ID: [email protected]:5060
  550. CSeq: 2151 BYE
  551. Server: Asterisk PBX 13.2.0
  552. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  553. Supported: replaces, timer
  554. Content-Length: 0
  555.  
  556.  
  557. <------------>
  558. -- Channel SIP/893-000000a1 left 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
  559. -- Channel SIP/892-000000a0 left 'simple_bridge' basic-bridge <3579d350-4183-4d03-b093-c4ec57b3690b>
  560. == Spawn extension (default, 893, 1) exited non-zero on 'SIP/892-000000a0'
  561. Scheduling destruction of SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' in 32000 ms (Method: INVITE)
  562. set_destination: Parsing <sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
  563. set_destination: URI is for WebSocket, we can't set destination
  564. Reliably Transmitting (no NAT) to 192.168.88.188:5060:
  565. BYE sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
  566. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK7ddb39ef
  567. Max-Forwards: 70
  568. From: <sip:[email protected]>;tag=as1a240a25
  569. To: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  570. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  571. CSeq: 102 BYE
  572. User-Agent: Asterisk PBX 13.2.0
  573. Proxy-Authorization: Digest username="889", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="1cded12e", response="06633f736457f7e9fcfe5d5359283a01"
  574. X-Asterisk-HangupCause: Normal Clearing
  575. X-Asterisk-HangupCauseCode: 16
  576. Content-Length: 0
  577.  
  578.  
  579. ---
  580.  
  581. <--- SIP read from WS:192.168.88.188:49766 --->
  582. SIP/2.0 200 OK
  583. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK7ddb39ef
  584. From: <sip:[email protected]>;tag=as1a240a25
  585. To: "892"<sip:[email protected]>;tag=WuTzQfG02C9oc1KGlJqA
  586. Contact: <sip:[email protected];transport=ws>
  587. Call-ID: 7252dd20-3201-26b9-e94f-1754d3b61c84
  588. CSeq: 102 BYE
  589. Content-Length: 0
  590.  
  591. <------------->
  592. --- (8 headers 0 lines) ---
  593. SIP Response message for INCOMING dialog BYE arrived
  594.  
  595. <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
  596. SIP/2.0 180 Ringing
  597. Via: SIP/2.0/WS 192.0.2.97;branch=z9hG4bK6600227;received=192.168.88.182
  598. From: <sip:[email protected]>;tag=ev8b7t1gfj
  599. To: <sip:[email protected]>;tag=as6bbeef1d
  600. Call-ID: s9iigcbrqmig8r9as7e7
  601. CSeq: 3015 INVITE
  602. Server: Asterisk PBX 13.2.0
  603. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  604. Supported: replaces, timer
  605. Contact: <sip:[email protected]:5060;transport=WS>
  606. Content-Length: 0
  607.  
  608.  
  609. <------------>
  610. Really destroying SIP dialog '7252dd20-3201-26b9-e94f-1754d3b61c84' Method: INVITE
  611. Really destroying SIP dialog '[email protected]:5060' Method: BYE
  612. ast*CLI> sip set debug off
  613. SIP Debugging Disabled
  614. ast*CLI>
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