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  1. Asterisk 15.0.0, Copyright (C) 1999 - 2016, Digium, Inc. and others.
  2. Created by Mark Spencer <markster@digium.com>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 15.0.0 currently running on Asty (pid = 6313)
  9. Asty*CLI> sip set debug on
  10. SIP Debugging enabled
  11.  
  12. <--- SIP read from TCP:10.11.1.100:45222 --->
  13. INVITE sip:1000@10.10.10.252:5060 SIP/2.0
  14. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.G2dfXKC2D;rport
  15. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  16. To: sip:1000@10.10.10.252:5060
  17. CSeq: 20 INVITE
  18. Call-ID: AfBhwWyFn-
  19. Max-Forwards: 70
  20. Supported: replaces, outbound
  21. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  22. Content-Type: application/sdp
  23. Content-Length: 498
  24. Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  25. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  26.  
  27. v=0
  28. o=bilal.lodhia 1154 3764 IN IP4 10.11.1.100
  29. s=Talk
  30. c=IN IP4 10.11.1.100
  31. t=0 0
  32. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  33. m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100
  34. a=rtpmap:96 opus/48000/2
  35. a=fmtp:96 useinbandfec=1
  36. a=rtpmap:97 speex/16000
  37. a=fmtp:97 vbr=on
  38. a=rtpmap:98 speex/8000
  39. a=fmtp:98 vbr=on
  40. a=fmtp:18 annexb=yes
  41. a=rtpmap:101 telephone-event/48000
  42. a=rtpmap:99 telephone-event/16000
  43. a=rtpmap:100 telephone-event/8000
  44. a=rtcp-fb:* ccm tmmbr
  45.  
  46. <------------->
  47. --- (13 headers 18 lines) ---
  48. Sending to 10.11.1.100:45222 (no NAT)
  49. Sending to 10.11.1.100:45222 (no NAT)
  50. Using INVITE request as basis request - AfBhwWyFn-
  51. Found peer 'bilal.lodhia' for 'bilal.lodhia' from 10.11.1.100:45222
  52.  
  53. <--- Reliably Transmitting (no NAT) to 10.11.1.100:45222 --->
  54. SIP/2.0 401 Unauthorized
  55. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.G2dfXKC2D;received=10.11.1.100;rport=45222
  56. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  57. To: sip:1000@10.10.10.252:5060;tag=as22a9d13c
  58. Call-ID: AfBhwWyFn-
  59. CSeq: 20 INVITE
  60. Server: Asterisk PBX 15.0.0
  61. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  62. Supported: replaces, timer
  63. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79ddd649"
  64. Content-Length: 0
  65.  
  66.  
  67. <------------>
  68. Scheduling destruction of SIP dialog 'AfBhwWyFn-' in 32000 ms (Method: INVITE)
  69.  
  70. <--- SIP read from TCP:10.11.1.100:45222 --->
  71. ACK sip:1000@10.10.10.252:5060 SIP/2.0
  72. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.G2dfXKC2D;rport
  73. Call-ID: AfBhwWyFn-
  74. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  75. To: <sip:1000@10.10.10.252:5060>;tag=as22a9d13c
  76. Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  77. Max-Forwards: 70
  78. CSeq: 20 ACK
  79. Content-Length: 0
  80.  
  81.  
  82. <------------->
  83. --- (9 headers 0 lines) ---
  84.  
  85. <--- SIP read from TCP:10.11.1.100:45222 --->
  86. INVITE sip:1000@10.10.10.252:5060 SIP/2.0
  87. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.4NJvvMTrc;rport
  88. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  89. To: sip:1000@10.10.10.252:5060
  90. CSeq: 21 INVITE
  91. Call-ID: AfBhwWyFn-
  92. Max-Forwards: 70
  93. Supported: replaces, outbound
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  95. Content-Type: application/sdp
  96. Content-Length: 498
  97. Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  98. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  99. Authorization: Digest realm="asterisk", nonce="79ddd649", algorithm=MD5, username="bilal.lodhia", uri="sip:1000@110.93.203.154", response="7754597ce21c36db4db7145c236576aa"
  100.  
  101. v=0
  102. o=bilal.lodhia 1154 3764 IN IP4 10.11.1.100
  103. s=Talk
  104. c=IN IP4 10.11.1.100
  105. t=0 0
  106. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  107. m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100
  108. a=rtpmap:96 opus/48000/2
  109. a=fmtp:96 useinbandfec=1
  110. a=rtpmap:97 speex/16000
  111. a=fmtp:97 vbr=on
  112. a=rtpmap:98 speex/8000
  113. a=fmtp:98 vbr=on
  114. a=fmtp:18 annexb=yes
  115. a=rtpmap:101 telephone-event/48000
  116. a=rtpmap:99 telephone-event/16000
  117. a=rtpmap:100 telephone-event/8000
  118. a=rtcp-fb:* ccm tmmbr
  119.  
  120. <------------->
  121. --- (14 headers 18 lines) ---
  122. Sending to 10.11.1.100:45222 (no NAT)
  123. Using INVITE request as basis request - AfBhwWyFn-
  124. Found peer 'bilal.lodhia' for 'bilal.lodhia' from 10.11.1.100:45222
  125. Found RTP audio format 96
  126. Found RTP audio format 97
  127. Found RTP audio format 98
  128. Found RTP audio format 0
  129. Found RTP audio format 8
  130. Found RTP audio format 18
  131. Found RTP audio format 101
  132. Found RTP audio format 99
  133. Found RTP audio format 100
  134. Found audio description format opus for ID 96
  135. Found audio description format speex for ID 97
  136. Found audio description format speex for ID 98
  137. Found unknown media description format telephone-event for ID 101
  138. Found unknown media description format telephone-event for ID 99
  139. Found audio description format telephone-event for ID 100
  140. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|alaw|g729|opus|speex16|speex)/video=(nothing)/text=(nothing), combined - (opus|speex16|ulaw)
  141. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  142. Peer audio RTP is at port 10.11.1.100:7076
  143. Looking for 1000 in common (domain 10.10.10.252)
  144. sip_route_dump: route/path hop: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>
  145.  
  146. <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
  147. SIP/2.0 100 Trying
  148. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.4NJvvMTrc;received=10.11.1.100;rport=45222
  149. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  150. To: sip:1000@10.10.10.252:5060
  151. Call-ID: AfBhwWyFn-
  152. CSeq: 21 INVITE
  153. Server: Asterisk PBX 15.0.0
  154. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  155. Supported: replaces, timer
  156. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  157. Content-Length: 0
  158.  
  159.  
  160. <------------>
  161. Audio is at 15620
  162. Adding codec opus to SDP
  163. Adding codec speex16 to SDP
  164. Adding codec speex32 to SDP
  165. Adding codec silk24 to SDP
  166. Adding codec silk16 to SDP
  167. Adding codec ulaw to SDP
  168. Adding non-codec 0x1 (telephone-event) to SDP
  169. Reliably Transmitting (no NAT) to 10.11.3.58:63611:
  170. INVITE sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
  171. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1184bc25
  172. Max-Forwards: 70
  173. From: <sip:bilal.lodhia@10.10.10.252>;tag=as59e28524
  174. To: <sip:God@10.11.3.58:63611;transport=tcp>
  175. Contact: <sip:bilal.lodhia@10.10.10.252:5060;transport=tcp>
  176. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  177. CSeq: 102 INVITE
  178. User-Agent: Asterisk PBX 15.0.0
  179. Date: Tue, 31 Oct 2017 03:11:04 GMT
  180. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  181. Supported: replaces, timer
  182. Content-Type: application/sdp
  183. Content-Length: 323
  184.  
  185. v=0
  186. o=root 617065444 617065444 IN IP4 10.10.10.252
  187. s=Asterisk PBX 15.0.0
  188. c=IN IP4 10.10.10.252
  189. t=0 0
  190. m=audio 15620 RTP/AVP 96 97 119 0 100
  191. a=rtpmap:96 opus/48000/2
  192. a=rtpmap:97 speex/16000
  193. a=rtpmap:119 speex/32000
  194. a=rtpmap:0 PCMU/8000
  195. a=rtpmap:100 telephone-event/8000
  196. a=fmtp:100 0-16
  197. a=maxptime:60
  198. a=sendrecv
  199.  
  200. ---
  201.  
  202. <--- SIP read from TCP:10.11.3.58:63611 --->
  203. SIP/2.0 100 Trying
  204. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1184bc25
  205. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  206. To: <sip:God@10.11.3.58:63611;transport=tcp>
  207. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  208. CSeq: 102 INVITE
  209. Content-Length: 0
  210.  
  211.  
  212. <------------->
  213. --- (7 headers 0 lines) ---
  214.  
  215. <--- SIP read from TCP:10.11.3.58:63611 --->
  216. SIP/2.0 180 Ringing
  217. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1184bc25
  218. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  219. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  220. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  221. CSeq: 102 INVITE
  222. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  223. Supported: replaces, outbound
  224. Content-Length: 0
  225.  
  226.  
  227. <------------->
  228. --- (9 headers 0 lines) ---
  229. sip_route_dump: no route/path
  230.  
  231. <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
  232. SIP/2.0 180 Ringing
  233. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.4NJvvMTrc;received=10.11.1.100;rport=45222
  234. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  235. To: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  236. Call-ID: AfBhwWyFn-
  237. CSeq: 21 INVITE
  238. Server: Asterisk PBX 15.0.0
  239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  240. Supported: replaces, timer
  241. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  242. Content-Length: 0
  243.  
  244.  
  245. <------------>
  246.  
  247. <--- SIP read from TCP:10.11.3.58:63611 --->
  248. SIP/2.0 200 Ok
  249. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1184bc25
  250. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  251. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  252. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  253. CSeq: 102 INVITE
  254. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  255. Supported: replaces, outbound
  256. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  257. Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  258. Content-Type: application/sdp
  259. Content-Length: 289
  260.  
  261. v=0
  262. o=God 1337 3518 IN IP4 10.11.3.58
  263. s=Talk
  264. c=IN IP4 10.11.3.58
  265. t=0 0
  266. m=audio 7078 RTP/AVP 96 97 119 0 100
  267. a=rtpmap:96 opus/48000/2
  268. a=fmtp:96 useinbandfec=1
  269. a=rtpmap:97 speex/16000
  270. a=fmtp:97 vbr=on
  271. a=rtpmap:119 speex/32000
  272. a=fmtp:119 vbr=on
  273. a=rtpmap:100 telephone-event/8000
  274.  
  275. <------------->
  276. --- (12 headers 13 lines) ---
  277. Found RTP audio format 96
  278. Found RTP audio format 97
  279. Found RTP audio format 119
  280. Found RTP audio format 0
  281. Found RTP audio format 100
  282. Found audio description format opus for ID 96
  283. Found audio description format speex for ID 97
  284. Found audio description format speex for ID 119
  285. Found audio description format telephone-event for ID 100
  286. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|opus|speex16|speex32)/video=(nothing)/text=(nothing), combined - (opus|speex16|speex32|ulaw)
  287. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  288. Peer audio RTP is at port 10.11.3.58:7078
  289. sip_route_dump: route/path hop: <sip:God@10.11.3.58:63611;transport=tcp>
  290. Transmitting (no NAT) to 10.11.3.58:63611:
  291. ACK sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
  292. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK2fc5bb91
  293. Max-Forwards: 70
  294. From: <sip:bilal.lodhia@10.10.10.252>;tag=as59e28524
  295. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  296. Contact: <sip:bilal.lodhia@10.10.10.252:5060;transport=tcp>
  297. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  298. CSeq: 102 ACK
  299. User-Agent: Asterisk PBX 15.0.0
  300. Content-Length: 0
  301.  
  302.  
  303. ---
  304. Audio is at 16118
  305. Adding codec opus to SDP
  306. Adding codec speex16 to SDP
  307. Adding codec ulaw to SDP
  308. Adding non-codec 0x1 (telephone-event) to SDP
  309.  
  310. <--- Reliably Transmitting (no NAT) to 10.11.1.100:45222 --->
  311. SIP/2.0 200 OK
  312. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.4NJvvMTrc;received=10.11.1.100;rport=45222
  313. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  314. To: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  315. Call-ID: AfBhwWyFn-
  316. CSeq: 21 INVITE
  317. Server: Asterisk PBX 15.0.0
  318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  319. Supported: replaces, timer
  320. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  321. Content-Type: application/sdp
  322. Content-Length: 295
  323.  
  324. v=0
  325. o=root 2014770230 2014770230 IN IP4 10.10.10.252
  326. s=Asterisk PBX 15.0.0
  327. c=IN IP4 10.10.10.252
  328. t=0 0
  329. m=audio 16118 RTP/AVP 96 97 0 100
  330. a=rtpmap:96 opus/48000/2
  331. a=rtpmap:97 speex/16000
  332. a=rtpmap:0 PCMU/8000
  333. a=rtpmap:100 telephone-event/8000
  334. a=fmtp:100 0-16
  335. a=maxptime:60
  336. a=sendrecv
  337.  
  338. <------------>
  339. Audio is at 15620
  340. Adding codec opus to SDP
  341. Adding codec speex16 to SDP
  342. Adding codec speex32 to SDP
  343. Adding codec silk24 to SDP
  344. Adding codec silk16 to SDP
  345. Adding codec ulaw to SDP
  346. Adding non-codec 0x1 (telephone-event) to SDP
  347. Reliably Transmitting (no NAT) to 10.11.3.58:63611:
  348. INVITE sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
  349. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK05d38a31
  350. Max-Forwards: 70
  351. From: <sip:bilal.lodhia@10.10.10.252>;tag=as59e28524
  352. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  353. Contact: <sip:bilal.lodhia@10.10.10.252:5060;transport=tcp>
  354. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  355. CSeq: 103 INVITE
  356. User-Agent: Asterisk PBX 15.0.0
  357. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  358. Supported: replaces, timer
  359. X-asterisk-Info: SIP re-invite (External RTP bridge)
  360. Content-Type: application/sdp
  361. Content-Length: 320
  362.  
  363. v=0
  364. o=root 617065444 617065445 IN IP4 10.11.1.100
  365. s=Asterisk PBX 15.0.0
  366. c=IN IP4 10.11.1.100
  367. t=0 0
  368. m=audio 7076 RTP/AVP 96 97 119 0 100
  369. a=rtpmap:96 opus/48000/2
  370. a=rtpmap:97 speex/16000
  371. a=rtpmap:119 speex/32000
  372. a=rtpmap:0 PCMU/8000
  373. a=rtpmap:100 telephone-event/8000
  374. a=fmtp:100 0-16
  375. a=maxptime:60
  376. a=sendrecv
  377.  
  378. ---
  379.  
  380. <--- SIP read from TCP:10.11.3.58:63611 --->
  381. SIP/2.0 100 Trying
  382. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK05d38a31
  383. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  384. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  385. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  386. CSeq: 103 INVITE
  387. Content-Length: 0
  388.  
  389.  
  390. <------------->
  391. --- (7 headers 0 lines) ---
  392.  
  393. <--- SIP read from TCP:10.11.3.58:63611 --->
  394. SIP/2.0 200 Ok
  395. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK05d38a31
  396. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  397. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  398. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  399. CSeq: 103 INVITE
  400. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  401. Supported: replaces, outbound
  402. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  403. Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  404. Content-Type: application/sdp
  405. Content-Length: 289
  406.  
  407. v=0
  408. o=God 1337 3520 IN IP4 10.11.3.58
  409. s=Talk
  410. c=IN IP4 10.11.3.58
  411. t=0 0
  412. m=audio 7078 RTP/AVP 96 97 119 0 100
  413. a=rtpmap:96 opus/48000/2
  414. a=fmtp:96 useinbandfec=1
  415. a=rtpmap:97 speex/16000
  416. a=fmtp:97 vbr=on
  417. a=rtpmap:119 speex/32000
  418. a=fmtp:119 vbr=on
  419. a=rtpmap:100 telephone-event/8000
  420.  
  421. <------------->
  422. --- (12 headers 13 lines) ---
  423. Found RTP audio format 96
  424. Found RTP audio format 97
  425. Found RTP audio format 119
  426. Found RTP audio format 0
  427. Found RTP audio format 100
  428. Found audio description format opus for ID 96
  429. Found audio description format speex for ID 97
  430. Found audio description format speex for ID 119
  431. Found audio description format telephone-event for ID 100
  432. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|opus|speex16|speex32)/video=(nothing)/text=(nothing), combined - (opus|speex16|speex32|ulaw)
  433. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  434. Peer audio RTP is at port 10.11.3.58:7078
  435. Transmitting (no NAT) to 10.11.3.58:63611:
  436. ACK sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
  437. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK568e1557
  438. Max-Forwards: 70
  439. From: <sip:bilal.lodhia@10.10.10.252>;tag=as59e28524
  440. To: <sip:God@10.11.3.58:63611;transport=tcp>;tag=V4hm-wW
  441. Contact: <sip:bilal.lodhia@10.10.10.252:5060;transport=tcp>
  442. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  443. CSeq: 103 ACK
  444. User-Agent: Asterisk PBX 15.0.0
  445. Content-Length: 0
  446.  
  447.  
  448. ---
  449.  
  450. <--- SIP read from TCP:10.11.1.100:45222 --->
  451. ACK sip:1000@10.10.10.252:5060;transport=tcp SIP/2.0
  452. Via: SIP/2.0/TCP 10.11.1.100:45222;rport;branch=z9hG4bK.OazurvlCn
  453. From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  454. To: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  455. CSeq: 21 ACK
  456. Call-ID: AfBhwWyFn-
  457. Max-Forwards: 70
  458. Authorization: Digest realm="asterisk", nonce="79ddd649", algorithm=MD5, username="bilal.lodhia", uri="sip:1000@110.93.203.154", response="7754597ce21c36db4db7145c236576aa"
  459. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  460. Content-Length: 0
  461.  
  462.  
  463. <------------->
  464. --- (10 headers 0 lines) ---
  465. Audio is at 16118
  466. Adding codec opus to SDP
  467. Adding non-codec 0x1 (telephone-event) to SDP
  468. Reliably Transmitting (no NAT) to 10.11.1.100:45222:
  469. INVITE sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
  470. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK400af390;rport
  471. Max-Forwards: 70
  472. From: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  473. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  474. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  475. Call-ID: AfBhwWyFn-
  476. CSeq: 102 INVITE
  477. User-Agent: Asterisk PBX 15.0.0
  478. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  479. Supported: replaces, timer
  480. X-asterisk-Info: SIP re-invite (External RTP bridge)
  481. Content-Type: application/sdp
  482. Content-Length: 238
  483.  
  484. v=0
  485. o=root 2014770230 2014770231 IN IP4 10.11.3.58
  486. s=Asterisk PBX 15.0.0
  487. c=IN IP4 10.11.3.58
  488. t=0 0
  489. m=audio 7078 RTP/AVP 96 100
  490. a=rtpmap:96 opus/48000/2
  491. a=rtpmap:100 telephone-event/8000
  492. a=fmtp:100 0-16
  493. a=maxptime:60
  494. a=sendrecv
  495.  
  496. ---
  497.  
  498. <--- SIP read from TCP:10.11.1.100:45222 --->
  499. SIP/2.0 100 Trying
  500. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK400af390;rport
  501. From: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  502. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  503. Call-ID: AfBhwWyFn-
  504. CSeq: 102 INVITE
  505. Content-Length: 0
  506.  
  507.  
  508. <------------->
  509. --- (7 headers 0 lines) ---
  510.  
  511. <--- SIP read from TCP:10.11.1.100:45222 --->
  512. SIP/2.0 200 Ok
  513. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK400af390;rport
  514. From: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  515. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  516. Call-ID: AfBhwWyFn-
  517. CSeq: 102 INVITE
  518. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  519. Supported: replaces, outbound
  520. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  521. Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  522. Content-Type: application/sdp
  523. Content-Length: 203
  524.  
  525. v=0
  526. o=bilal.lodhia 1154 3766 IN IP4 10.11.1.100
  527. s=Talk
  528. c=IN IP4 10.11.1.100
  529. t=0 0
  530. m=audio 7076 RTP/AVP 96 100
  531. a=rtpmap:96 opus/48000/2
  532. a=fmtp:96 useinbandfec=1
  533. a=rtpmap:100 telephone-event/8000
  534.  
  535. <------------->
  536. --- (12 headers 9 lines) ---
  537. Found RTP audio format 96
  538. Found RTP audio format 100
  539. Found audio description format opus for ID 96
  540. Found audio description format telephone-event for ID 100
  541. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
  542. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  543. Peer audio RTP is at port 10.11.1.100:7076
  544. Transmitting (no NAT) to 10.11.1.100:45222:
  545. ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
  546. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK7db367a8;rport
  547. Max-Forwards: 70
  548. From: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  549. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  550. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  551. Call-ID: AfBhwWyFn-
  552. CSeq: 102 ACK
  553. User-Agent: Asterisk PBX 15.0.0
  554. Content-Length: 0
  555.  
  556.  
  557. ---
  558.  
  559. <--- SIP read from TCP:10.11.3.58:63611 --->
  560. BYE sip:bilal.lodhia@10.10.10.252:5060;transport=tcp SIP/2.0
  561. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.12qpcdwps;rport
  562. From: <sip:God@10.11.3.58>;tag=V4hm-wW
  563. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  564. CSeq: 111 BYE
  565. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  566. Max-Forwards: 70
  567. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  568. Content-Length: 0
  569.  
  570.  
  571. <------------->
  572. --- (9 headers 0 lines) ---
  573. Sending to 10.11.3.58:63611 (no NAT)
  574. Scheduling destruction of SIP dialog '72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060' in 32000 ms (Method: BYE)
  575.  
  576. <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
  577. SIP/2.0 200 OK
  578. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.12qpcdwps;received=10.11.3.58;rport=63611
  579. From: <sip:God@10.11.3.58>;tag=V4hm-wW
  580. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=as59e28524
  581. Call-ID: 72af47fb4e7aee447b406dbd1e4d14fc@10.10.10.252:5060
  582. CSeq: 111 BYE
  583. Server: Asterisk PBX 15.0.0
  584. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  585. Supported: replaces, timer
  586. Content-Length: 0
  587.  
  588.  
  589. <------------>
  590. Audio is at 16118
  591. Adding codec opus to SDP
  592. Adding non-codec 0x1 (telephone-event) to SDP
  593. Reliably Transmitting (no NAT) to 10.11.1.100:45222:
  594. INVITE sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
  595. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1bc639c1;rport
  596. Max-Forwards: 70
  597. From: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  598. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  599. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  600. Call-ID: AfBhwWyFn-
  601. CSeq: 103 INVITE
  602. User-Agent: Asterisk PBX 15.0.0
  603. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  604. Supported: replaces, timer
  605. X-asterisk-Info: SIP re-invite (External RTP bridge)
  606. Content-Type: application/sdp
  607. Content-Length: 243
  608.  
  609. v=0
  610. o=root 2014770230 2014770232 IN IP4 10.10.10.252
  611. s=Asterisk PBX 15.0.0
  612. c=IN IP4 10.10.10.252
  613. t=0 0
  614. m=audio 16118 RTP/AVP 96 100
  615. a=rtpmap:96 opus/48000/2
  616. a=rtpmap:100 telephone-event/8000
  617. a=fmtp:100 0-16
  618. a=maxptime:60
  619. a=sendrecv
  620.  
  621. ---
  622. Scheduling destruction of SIP dialog 'AfBhwWyFn-' in 32000 ms (Method: ACK)
  623.  
  624. <--- SIP read from TCP:10.11.1.100:45222 --->
  625. SIP/2.0 100 Trying
  626. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1bc639c1;rport
  627. From: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  628. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  629. Call-ID: AfBhwWyFn-
  630. CSeq: 103 INVITE
  631. Content-Length: 0
  632.  
  633.  
  634. <------------->
  635. --- (7 headers 0 lines) ---
  636.  
  637. <--- SIP read from TCP:10.11.1.100:45222 --->
  638. SIP/2.0 200 Ok
  639. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK1bc639c1;rport
  640. From: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  641. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  642. Call-ID: AfBhwWyFn-
  643. CSeq: 103 INVITE
  644. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  645. Supported: replaces, outbound
  646. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  647. Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  648. Content-Type: application/sdp
  649. Content-Length: 203
  650.  
  651. v=0
  652. o=bilal.lodhia 1154 3768 IN IP4 10.11.1.100
  653. s=Talk
  654. c=IN IP4 10.11.1.100
  655. t=0 0
  656. m=audio 7076 RTP/AVP 96 100
  657. a=rtpmap:96 opus/48000/2
  658. a=fmtp:96 useinbandfec=1
  659. a=rtpmap:100 telephone-event/8000
  660.  
  661. <------------->
  662. --- (12 headers 9 lines) ---
  663. Found RTP audio format 96
  664. Found RTP audio format 100
  665. Found audio description format opus for ID 96
  666. Found audio description format telephone-event for ID 100
  667. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
  668. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  669. Peer audio RTP is at port 10.11.1.100:7076
  670. Transmitting (no NAT) to 10.11.1.100:45222:
  671. ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
  672. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK577bbd96;rport
  673. Max-Forwards: 70
  674. From: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  675. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  676. Contact: <sip:1000@10.10.10.252:5060;transport=tcp>
  677. Call-ID: AfBhwWyFn-
  678. CSeq: 103 ACK
  679. User-Agent: Asterisk PBX 15.0.0
  680. Content-Length: 0
  681.  
  682.  
  683. ---
  684. Reliably Transmitting (no NAT) to 10.11.1.100:45222:
  685. BYE sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
  686. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6959913c;rport
  687. Max-Forwards: 70
  688. From: sip:1000@10.10.10.252:5060;tag=as6bd52a2a
  689. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  690. Call-ID: AfBhwWyFn-
  691. CSeq: 104 BYE
  692. User-Agent: Asterisk PBX 15.0.0
  693. Proxy-Authorization: Digest username="bilal.lodhia", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.252", nonce="79ddd649", response="028f33d093722b95832954705c404b7e"
  694. X-Asterisk-HangupCause: Normal Clearing
  695. X-Asterisk-HangupCauseCode: 16
  696. Content-Length: 0
  697.  
  698.  
  699. ---
  700. Scheduling destruction of SIP dialog 'AfBhwWyFn-' in 32000 ms (Method: ACK)
  701.  
  702. <--- SIP read from TCP:10.11.1.100:45222 --->
  703. SIP/2.0 200 Ok
  704. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6959913c;rport
  705. From: <sip:1000@10.10.10.252:5060>;tag=as6bd52a2a
  706. To: <sip:bilal.lodhia@10.10.10.252:5060>;tag=uHQnqBc7N
  707. Call-ID: AfBhwWyFn-
  708. CSeq: 104 BYE
  709. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  710. Supported: replaces, outbound
  711. Content-Length: 0
  712.  
  713.  
  714. <------------->
  715. --- (9 headers 0 lines) ---
  716. Really destroying SIP dialog 'AfBhwWyFn-' Method: ACK
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