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  1. Asterisk 15.0.0, Copyright (C) 1999 - 2016, Digium, Inc. and others.
  2. Created by Mark Spencer <[email protected]>
  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:10.11.3.58:63611 --->
  13. INVITE sip:[email protected] SIP/2.0
  14. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;rport
  15. From: <sip:[email protected]>;tag=WGTTVGU85
  16. To: sip:[email protected]
  17. CSeq: 20 INVITE
  18. Call-ID: UYTfLSiokR
  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: 954
  24. Contact: <sip:[email protected]:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  25. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  26.  
  27. v=0
  28. o=God 1019 1492 IN IP4 10.11.3.58
  29. s=Talk
  30. c=IN IP4 10.11.3.58
  31. t=0 0
  32. a=ice-pwd:0f022f7358cd64f6c4450f9a
  33. a=ice-ufrag:6947d51f
  34. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  35. m=audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
  36. c=IN IP4 110.93.203.154
  37. a=rtpmap:96 opus/48000/2
  38. a=fmtp:96 useinbandfec=1
  39. a=rtpmap:97 speex/16000
  40. a=fmtp:97 vbr=on
  41. a=rtpmap:98 speex/8000
  42. a=fmtp:98 vbr=on
  43. a=rtpmap:99 speex/32000
  44. a=fmtp:99 vbr=on
  45. a=rtpmap:101 telephone-event/48000
  46. a=rtpmap:100 telephone-event/16000
  47. a=rtpmap:102 telephone-event/8000
  48. a=rtpmap:103 telephone-event/32000
  49. a=candidate:1 1 UDP 2130706431 10.11.3.58 7078 typ host
  50. a=candidate:1 2 UDP 2130706430 10.11.3.58 7079 typ host
  51. a=candidate:2 1 UDP 1694498815 110.93.203.154 7078 typ srflx raddr 10.11.3.58 rport 7078
  52. a=candidate:2 2 UDP 1694498814 110.93.203.154 7079 typ srflx raddr 10.11.3.58 rport 7079
  53. a=rtcp-fb:* trr-int 5000
  54. a=rtcp-fb:* ccm tmmbr
  55.  
  56. <------------->
  57. --- (13 headers 28 lines) ---
  58. Sending to 10.11.3.58:63611 (no NAT)
  59. Sending to 10.11.3.58:63611 (no NAT)
  60. Using INVITE request as basis request - UYTfLSiokR
  61. Found peer 'God' for 'God' from 10.11.3.58:63611
  62.  
  63. <--- Reliably Transmitting (no NAT) to 10.11.3.58:63611 --->
  64. SIP/2.0 401 Unauthorized
  65. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;received=10.11.3.58;rport=63611
  66. From: <sip:[email protected]>;tag=WGTTVGU85
  67. To: sip:[email protected];tag=as09232fff
  68. Call-ID: UYTfLSiokR
  69. CSeq: 20 INVITE
  70. Server: Asterisk PBX 15.0.0
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  72. Supported: replaces, timer
  73. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59fb78f5"
  74. Content-Length: 0
  75.  
  76.  
  77. <------------>
  78. Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: INVITE)
  79.  
  80. <--- SIP read from TCP:10.11.3.58:63611 --->
  81. ACK sip:[email protected]:5060 SIP/2.0
  82. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;rport
  83. Call-ID: UYTfLSiokR
  84. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  85. To: <sip:[email protected]:5060>;tag=as09232fff
  86. Contact: <sip:[email protected]:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  87. Max-Forwards: 70
  88. CSeq: 20 ACK
  89. Content-Length: 0
  90.  
  91.  
  92. <------------->
  93. --- (9 headers 0 lines) ---
  94.  
  95. <--- SIP read from TCP:10.11.3.58:63611 --->
  96. INVITE sip:[email protected] SIP/2.0
  97. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;rport
  98. From: <sip:[email protected]>;tag=WGTTVGU85
  99. To: sip:[email protected]
  100. CSeq: 21 INVITE
  101. Call-ID: UYTfLSiokR
  102. Max-Forwards: 70
  103. Supported: replaces, outbound
  104. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  105. Content-Type: application/sdp
  106. Content-Length: 954
  107. Contact: <sip:[email protected]:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  108. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  109. Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:[email protected]", response="b0de4823a7a50733150b148e2d0fb8c0"
  110.  
  111. v=0
  112. o=God 1019 1492 IN IP4 10.11.3.58
  113. s=Talk
  114. c=IN IP4 10.11.3.58
  115. t=0 0
  116. a=ice-pwd:0f022f7358cd64f6c4450f9a
  117. a=ice-ufrag:6947d51f
  118. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  119. m=audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
  120. c=IN IP4 110.93.203.154
  121. a=rtpmap:96 opus/48000/2
  122. a=fmtp:96 useinbandfec=1
  123. a=rtpmap:97 speex/16000
  124. a=fmtp:97 vbr=on
  125. a=rtpmap:98 speex/8000
  126. a=fmtp:98 vbr=on
  127. a=rtpmap:99 speex/32000
  128. a=fmtp:99 vbr=on
  129. a=rtpmap:101 telephone-event/48000
  130. a=rtpmap:100 telephone-event/16000
  131. a=rtpmap:102 telephone-event/8000
  132. a=rtpmap:103 telephone-event/32000
  133. a=candidate:1 1 UDP 2130706431 10.11.3.58 7078 typ host
  134. a=candidate:1 2 UDP 2130706430 10.11.3.58 7079 typ host
  135. a=candidate:2 1 UDP 1694498815 110.93.203.154 7078 typ srflx raddr 10.11.3.58 rport 7078
  136. a=candidate:2 2 UDP 1694498814 110.93.203.154 7079 typ srflx raddr 10.11.3.58 rport 7079
  137. a=rtcp-fb:* trr-int 5000
  138. a=rtcp-fb:* ccm tmmbr
  139.  
  140. <------------->
  141. --- (14 headers 28 lines) ---
  142. Sending to 10.11.3.58:63611 (no NAT)
  143. Using INVITE request as basis request - UYTfLSiokR
  144. Found peer 'God' for 'God' from 10.11.3.58:63611
  145. [Oct 31 08:08:02] NOTICE[7067][C-00011555]: chan_sip.c:10421 process_sdp: Received AVPF profile in audio offer but AVPF is not enabled, enabling: audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
  146. Found RTP audio format 96
  147. Found RTP audio format 97
  148. Found RTP audio format 98
  149. Found RTP audio format 0
  150. Found RTP audio format 8
  151. Found RTP audio format 99
  152. Found RTP audio format 101
  153. Found RTP audio format 100
  154. Found RTP audio format 102
  155. Found RTP audio format 103
  156. Found audio description format opus for ID 96
  157. Found audio description format speex for ID 97
  158. Found audio description format speex for ID 98
  159. Found audio description format speex for ID 99
  160. Found unknown media description format telephone-event for ID 101
  161. Found unknown media description format telephone-event for ID 100
  162. Found audio description format telephone-event for ID 102
  163. Found unknown media description format telephone-event for ID 103
  164. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|alaw|opus|speex16|speex|speex32)/video=(nothing)/text=(nothing), combined - (opus|speex16|speex32|ulaw)
  165. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  166. Peer audio RTP is at port 110.93.203.154:7078
  167. Looking for 1003 in common (domain 110.93.203.154)
  168. sip_route_dump: route/path hop: <sip:[email protected]:63611;transport=tcp>
  169.  
  170. <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
  171. SIP/2.0 100 Trying
  172. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
  173. From: <sip:[email protected]>;tag=WGTTVGU85
  174. To: sip:[email protected]
  175. Call-ID: UYTfLSiokR
  176. CSeq: 21 INVITE
  177. Server: Asterisk PBX 15.0.0
  178. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  179. Supported: replaces, timer
  180. Contact: <sip:[email protected]:5060;transport=tcp>
  181. Content-Length: 0
  182.  
  183.  
  184. <------------>
  185. Audio is at 10928
  186. Adding codec opus to SDP
  187. Adding codec speex16 to SDP
  188. Adding codec speex32 to SDP
  189. Adding codec silk24 to SDP
  190. Adding codec silk16 to SDP
  191. Adding codec ulaw to SDP
  192. Adding non-codec 0x1 (telephone-event) to SDP
  193. Reliably Transmitting (no NAT) to 10.11.1.100:45222:
  194. INVITE sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp SIP/2.0
  195. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  196. Max-Forwards: 70
  197. From: <sip:[email protected]>;tag=as11d0abee
  198. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>
  199. Contact: <sip:[email protected]:5060;transport=tcp>
  200. Call-ID: [email protected]:5060
  201. CSeq: 102 INVITE
  202. User-Agent: Asterisk PBX 15.0.0
  203. Date: Tue, 31 Oct 2017 03:08:02 GMT
  204. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  205. Supported: replaces, timer
  206. Content-Type: application/sdp
  207. Content-Length: 321
  208.  
  209. v=0
  210. o=root 633427397 633427397 IN IP4 10.10.10.252
  211. s=Asterisk PBX 15.0.0
  212. c=IN IP4 10.10.10.252
  213. t=0 0
  214. m=audio 10928 RTP/AVP 96 97 99 0 102
  215. a=rtpmap:96 opus/48000/2
  216. a=rtpmap:97 speex/16000
  217. a=rtpmap:99 speex/32000
  218. a=rtpmap:0 PCMU/8000
  219. a=rtpmap:102 telephone-event/8000
  220. a=fmtp:102 0-16
  221. a=maxptime:60
  222. a=sendrecv
  223.  
  224. ---
  225.  
  226. <--- SIP read from TCP:10.11.1.100:45222 --->
  227. SIP/2.0 100 Trying
  228. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  229. From: <sip:[email protected]:5060>;tag=as11d0abee
  230. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>
  231. Call-ID: [email protected]:5060
  232. CSeq: 102 INVITE
  233. Content-Length: 0
  234.  
  235.  
  236. <------------->
  237. --- (7 headers 0 lines) ---
  238.  
  239. <--- SIP read from TCP:10.11.1.100:45222 --->
  240. SIP/2.0 180 Ringing
  241. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  242. From: <sip:[email protected]:5060>;tag=as11d0abee
  243. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  244. Call-ID: [email protected]:5060
  245. CSeq: 102 INVITE
  246. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  247. Supported: replaces, outbound
  248. Content-Length: 0
  249.  
  250.  
  251. <------------->
  252. --- (9 headers 0 lines) ---
  253. sip_route_dump: no route/path
  254.  
  255. <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
  256. SIP/2.0 180 Ringing
  257. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
  258. From: <sip:[email protected]>;tag=WGTTVGU85
  259. To: sip:[email protected];tag=as5f4d7d62
  260. Call-ID: UYTfLSiokR
  261. CSeq: 21 INVITE
  262. Server: Asterisk PBX 15.0.0
  263. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  264. Supported: replaces, timer
  265. Contact: <sip:[email protected]:5060;transport=tcp>
  266. Content-Length: 0
  267.  
  268.  
  269. <------------>
  270. Really destroying SIP dialog 'mMhVlrBjlh' Method: REGISTER
  271.  
  272. <--- SIP read from TCP:10.11.1.100:45222 --->
  273. SIP/2.0 200 Ok
  274. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  275. From: <sip:[email protected]:5060>;tag=as11d0abee
  276. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  277. Call-ID: [email protected]:5060
  278. CSeq: 102 INVITE
  279. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  280. Supported: replaces, outbound
  281. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  282. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  283. Content-Type: application/sdp
  284. Content-Length: 251
  285.  
  286. v=0
  287. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  288. s=Talk
  289. c=IN IP4 10.11.1.100
  290. t=0 0
  291. m=audio 7076 RTP/AVP 96 97 0 102
  292. a=rtpmap:96 opus/48000/2
  293. a=fmtp:96 useinbandfec=1
  294. a=rtpmap:97 speex/16000
  295. a=fmtp:97 vbr=on
  296. a=rtpmap:102 telephone-event/8000
  297.  
  298. <------------->
  299. --- (12 headers 11 lines) ---
  300. Found RTP audio format 96
  301. Found RTP audio format 97
  302. Found RTP audio format 0
  303. Found RTP audio format 102
  304. Found audio description format opus for ID 96
  305. Found audio description format speex for ID 97
  306. Found audio description format telephone-event for ID 102
  307. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|opus|speex16)/video=(nothing)/text=(nothing), combined - (opus|speex16|ulaw)
  308. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  309. Peer audio RTP is at port 10.11.1.100:7076
  310. sip_route_dump: route/path hop: <sip:[email protected]:45222;transport=tcp>
  311. Transmitting (no NAT) to 10.11.1.100:45222:
  312. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  313. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3b7a684d
  314. Max-Forwards: 70
  315. From: <sip:[email protected]>;tag=as11d0abee
  316. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  317. Contact: <sip:[email protected]:5060;transport=tcp>
  318. Call-ID: [email protected]:5060
  319. CSeq: 102 ACK
  320. User-Agent: Asterisk PBX 15.0.0
  321. Content-Length: 0
  322.  
  323.  
  324. ---
  325. Audio is at 11938
  326. Adding codec opus to SDP
  327. Adding codec speex16 to SDP
  328. Adding codec speex32 to SDP
  329. Adding codec ulaw to SDP
  330. Adding non-codec 0x1 (telephone-event) to SDP
  331.  
  332. <--- Reliably Transmitting (no NAT) to 10.11.3.58:63611 --->
  333. SIP/2.0 200 OK
  334. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
  335. From: <sip:[email protected]>;tag=WGTTVGU85
  336. To: sip:[email protected];tag=as5f4d7d62
  337. Call-ID: UYTfLSiokR
  338. CSeq: 21 INVITE
  339. Server: Asterisk PBX 15.0.0
  340. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  341. Supported: replaces, timer
  342. Contact: <sip:[email protected]:5060;transport=tcp>
  343. Content-Type: application/sdp
  344. Content-Length: 324
  345.  
  346. v=0
  347. o=root 1076832751 1076832751 IN IP4 10.10.10.252
  348. s=Asterisk PBX 15.0.0
  349. c=IN IP4 10.10.10.252
  350. t=0 0
  351. m=audio 11938 RTP/AVPF 96 97 99 0 102
  352. a=rtpmap:96 opus/48000/2
  353. a=rtpmap:97 speex/16000
  354. a=rtpmap:99 speex/32000
  355. a=rtpmap:0 PCMU/8000
  356. a=rtpmap:102 telephone-event/8000
  357. a=fmtp:102 0-16
  358. a=maxptime:60
  359. a=sendrecv
  360.  
  361. <------------>
  362. Audio is at 10928
  363. Adding codec opus to SDP
  364. Adding codec speex16 to SDP
  365. Adding codec speex32 to SDP
  366. Adding codec silk24 to SDP
  367. Adding codec silk16 to SDP
  368. Adding codec ulaw to SDP
  369. Adding non-codec 0x1 (telephone-event) to SDP
  370. Reliably Transmitting (no NAT) to 10.11.1.100:45222:
  371. INVITE sip:[email protected]:45222;transport=tcp SIP/2.0
  372. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3ef7d81d
  373. Max-Forwards: 70
  374. From: <sip:[email protected]>;tag=as11d0abee
  375. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  376. Contact: <sip:[email protected]:5060;transport=tcp>
  377. Call-ID: [email protected]:5060
  378. CSeq: 103 INVITE
  379. User-Agent: Asterisk PBX 15.0.0
  380. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  381. Supported: replaces, timer
  382. X-asterisk-Info: SIP re-invite (External RTP bridge)
  383. Content-Type: application/sdp
  384. Content-Length: 324
  385.  
  386. v=0
  387. o=root 633427397 633427398 IN IP4 110.93.203.154
  388. s=Asterisk PBX 15.0.0
  389. c=IN IP4 110.93.203.154
  390. t=0 0
  391. m=audio 7078 RTP/AVP 96 97 99 0 102
  392. a=rtpmap:96 opus/48000/2
  393. a=rtpmap:97 speex/16000
  394. a=rtpmap:99 speex/32000
  395. a=rtpmap:0 PCMU/8000
  396. a=rtpmap:102 telephone-event/8000
  397. a=fmtp:102 0-16
  398. a=maxptime:60
  399. a=sendrecv
  400.  
  401. ---
  402.  
  403. <--- SIP read from TCP:10.11.3.58:63611 --->
  404. ACK sip:[email protected]:5060;transport=tcp SIP/2.0
  405. Via: SIP/2.0/TCP 10.11.3.58:63611;rport;branch=z9hG4bK.jhhHRGJBd
  406. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  407. To: <sip:[email protected]:5060>;tag=as5f4d7d62
  408. CSeq: 21 ACK
  409. Call-ID: UYTfLSiokR
  410. Max-Forwards: 70
  411. Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:[email protected]", response="b0de4823a7a50733150b148e2d0fb8c0"
  412. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  413. Content-Length: 0
  414.  
  415.  
  416. <------------->
  417. --- (10 headers 0 lines) ---
  418.  
  419. <--- SIP read from TCP:10.11.1.100:45222 --->
  420. SIP/2.0 400 Bad request
  421. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3b7a684d
  422. From: <sip:[email protected]:5060>;tag=as11d0abee
  423. Call-ID: [email protected]:5060
  424. CSeq: 102 ACK
  425. Content-Length: 0
  426.  
  427.  
  428. <------------->
  429. --- (6 headers 0 lines) ---
  430.  
  431. <--- SIP read from TCP:10.11.1.100:45222 --->
  432. SIP/2.0 400 Bad request
  433. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3ef7d81d
  434. From: <sip:[email protected]:5060>;tag=as11d0abee
  435. Call-ID: [email protected]:5060
  436. CSeq: 103 INVITE
  437. Content-Length: 0
  438.  
  439.  
  440. <------------->
  441. --- (6 headers 0 lines) ---
  442.  
  443. <--- SIP read from TCP:10.11.1.100:45222 --->
  444. SIP/2.0 200 Ok
  445. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  446. From: <sip:[email protected]:5060>;tag=as11d0abee
  447. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  448. Call-ID: [email protected]:5060
  449. CSeq: 102 INVITE
  450. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  451. Supported: replaces, outbound
  452. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  453. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  454. Content-Type: application/sdp
  455. Content-Length: 251
  456.  
  457. v=0
  458. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  459. s=Talk
  460. c=IN IP4 10.11.1.100
  461. t=0 0
  462. m=audio 7076 RTP/AVP 96 97 0 102
  463. a=rtpmap:96 opus/48000/2
  464. a=fmtp:96 useinbandfec=1
  465. a=rtpmap:97 speex/16000
  466. a=fmtp:97 vbr=on
  467. a=rtpmap:102 telephone-event/8000
  468.  
  469. <------------->
  470. --- (12 headers 11 lines) ---
  471. Transmitting (no NAT) to 10.11.1.100:45222:
  472. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  473. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4ed75a23
  474. Max-Forwards: 70
  475. From: <sip:[email protected]>;tag=as11d0abee
  476. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  477. Contact: <sip:[email protected]:5060;transport=tcp>
  478. Call-ID: [email protected]:5060
  479. CSeq: 102 ACK
  480. User-Agent: Asterisk PBX 15.0.0
  481. Content-Length: 0
  482.  
  483.  
  484. ---
  485.  
  486. <--- SIP read from TCP:10.11.1.100:45222 --->
  487. SIP/2.0 400 Bad request
  488. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4ed75a23
  489. From: <sip:[email protected]:5060>;tag=as11d0abee
  490. Call-ID: [email protected]:5060
  491. CSeq: 102 ACK
  492. Content-Length: 0
  493.  
  494.  
  495. <------------->
  496. --- (6 headers 0 lines) ---
  497. Audio is at 11938
  498. Adding codec opus to SDP
  499. Adding non-codec 0x1 (telephone-event) to SDP
  500. Reliably Transmitting (no NAT) to 10.11.3.58:63611:
  501. INVITE sip:[email protected]:63611;transport=tcp SIP/2.0
  502. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
  503. Max-Forwards: 70
  504. From: sip:[email protected];tag=as5f4d7d62
  505. To: <sip:[email protected]>;tag=WGTTVGU85
  506. Contact: <sip:[email protected]:5060;transport=tcp>
  507. Call-ID: UYTfLSiokR
  508. CSeq: 102 INVITE
  509. User-Agent: Asterisk PBX 15.0.0
  510. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  511. Supported: replaces, timer
  512. X-asterisk-Info: SIP re-invite (External RTP bridge)
  513. Content-Type: application/sdp
  514. Content-Length: 241
  515.  
  516. v=0
  517. o=root 1076832751 1076832752 IN IP4 10.11.1.100
  518. s=Asterisk PBX 15.0.0
  519. c=IN IP4 10.11.1.100
  520. t=0 0
  521. m=audio 7076 RTP/AVPF 96 102
  522. a=rtpmap:96 opus/48000/2
  523. a=rtpmap:102 telephone-event/8000
  524. a=fmtp:102 0-16
  525. a=maxptime:60
  526. a=sendrecv
  527.  
  528. ---
  529.  
  530. <--- SIP read from TCP:10.11.3.58:63611 --->
  531. SIP/2.0 100 Trying
  532. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
  533. From: <sip:[email protected]:5060>;tag=as5f4d7d62
  534. To: <sip:[email protected]:5060>;tag=WGTTVGU85
  535. Call-ID: UYTfLSiokR
  536. CSeq: 102 INVITE
  537. Content-Length: 0
  538.  
  539.  
  540. <------------->
  541. --- (7 headers 0 lines) ---
  542.  
  543. <--- SIP read from TCP:10.11.3.58:63611 --->
  544. SIP/2.0 200 Ok
  545. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
  546. From: <sip:[email protected]:5060>;tag=as5f4d7d62
  547. To: <sip:[email protected]:5060>;tag=WGTTVGU85
  548. Call-ID: UYTfLSiokR
  549. CSeq: 102 INVITE
  550. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  551. Supported: replaces, outbound
  552. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  553. Contact: <sip:[email protected]:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  554. Content-Type: application/sdp
  555. Content-Length: 219
  556.  
  557. v=0
  558. o=God 1019 1494 IN IP4 10.11.3.58
  559. s=Talk
  560. c=IN IP4 10.11.3.58
  561. t=0 0
  562. m=audio 7078 RTP/AVPF 96 102
  563. a=rtpmap:96 opus/48000/2
  564. a=fmtp:96 useinbandfec=1
  565. a=rtpmap:102 telephone-event/8000
  566. a=rtcp-fb:* trr-int 5000
  567.  
  568. <------------->
  569. --- (12 headers 10 lines) ---
  570. Found RTP audio format 96
  571. Found RTP audio format 102
  572. Found audio description format opus for ID 96
  573. Found audio description format telephone-event for ID 102
  574. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
  575. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  576. Peer audio RTP is at port 10.11.3.58:7078
  577. Transmitting (no NAT) to 10.11.3.58:63611:
  578. ACK sip:[email protected]:63611;transport=tcp SIP/2.0
  579. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK0f6dfe1a;rport
  580. Max-Forwards: 70
  581. From: sip:[email protected];tag=as5f4d7d62
  582. To: <sip:[email protected]>;tag=WGTTVGU85
  583. Contact: <sip:[email protected]:5060;transport=tcp>
  584. Call-ID: UYTfLSiokR
  585. CSeq: 102 ACK
  586. User-Agent: Asterisk PBX 15.0.0
  587. Content-Length: 0
  588.  
  589.  
  590. ---
  591.  
  592. <--- SIP read from TCP:10.11.1.100:45222 --->
  593. SIP/2.0 200 Ok
  594. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  595. From: <sip:[email protected]:5060>;tag=as11d0abee
  596. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  597. Call-ID: [email protected]:5060
  598. CSeq: 102 INVITE
  599. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  600. Supported: replaces, outbound
  601. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  602. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  603. Content-Type: application/sdp
  604. Content-Length: 251
  605.  
  606. v=0
  607. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  608. s=Talk
  609. c=IN IP4 10.11.1.100
  610. t=0 0
  611. m=audio 7076 RTP/AVP 96 97 0 102
  612. a=rtpmap:96 opus/48000/2
  613. a=fmtp:96 useinbandfec=1
  614. a=rtpmap:97 speex/16000
  615. a=fmtp:97 vbr=on
  616. a=rtpmap:102 telephone-event/8000
  617.  
  618. <------------->
  619. --- (12 headers 11 lines) ---
  620. Transmitting (no NAT) to 10.11.1.100:45222:
  621. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  622. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK54318e30
  623. Max-Forwards: 70
  624. From: <sip:[email protected]>;tag=as11d0abee
  625. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  626. Contact: <sip:[email protected]:5060;transport=tcp>
  627. Call-ID: [email protected]:5060
  628. CSeq: 102 ACK
  629. User-Agent: Asterisk PBX 15.0.0
  630. Content-Length: 0
  631.  
  632.  
  633. ---
  634.  
  635. <--- SIP read from TCP:10.11.1.100:45222 --->
  636. SIP/2.0 400 Bad request
  637. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK54318e30
  638. From: <sip:[email protected]:5060>;tag=as11d0abee
  639. Call-ID: [email protected]:5060
  640. CSeq: 102 ACK
  641. Content-Length: 0
  642.  
  643.  
  644. <------------->
  645. --- (6 headers 0 lines) ---
  646.  
  647. <--- SIP read from TCP:10.11.1.100:45222 --->
  648. SIP/2.0 200 Ok
  649. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  650. From: <sip:[email protected]:5060>;tag=as11d0abee
  651. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  652. Call-ID: [email protected]:5060
  653. CSeq: 102 INVITE
  654. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  655. Supported: replaces, outbound
  656. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  657. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  658. Content-Type: application/sdp
  659. Content-Length: 251
  660.  
  661. v=0
  662. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  663. s=Talk
  664. c=IN IP4 10.11.1.100
  665. t=0 0
  666. m=audio 7076 RTP/AVP 96 97 0 102
  667. a=rtpmap:96 opus/48000/2
  668. a=fmtp:96 useinbandfec=1
  669. a=rtpmap:97 speex/16000
  670. a=fmtp:97 vbr=on
  671. a=rtpmap:102 telephone-event/8000
  672.  
  673. <------------->
  674. --- (12 headers 11 lines) ---
  675. Transmitting (no NAT) to 10.11.1.100:45222:
  676. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  677. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6cb9f866
  678. Max-Forwards: 70
  679. From: <sip:[email protected]>;tag=as11d0abee
  680. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  681. Contact: <sip:[email protected]:5060;transport=tcp>
  682. Call-ID: [email protected]:5060
  683. CSeq: 102 ACK
  684. User-Agent: Asterisk PBX 15.0.0
  685. Content-Length: 0
  686.  
  687.  
  688. ---
  689.  
  690. <--- SIP read from TCP:10.11.1.100:45222 --->
  691. SIP/2.0 400 Bad request
  692. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6cb9f866
  693. From: <sip:[email protected]:5060>;tag=as11d0abee
  694. Call-ID: [email protected]:5060
  695. CSeq: 102 ACK
  696. Content-Length: 0
  697.  
  698.  
  699. <------------->
  700. --- (6 headers 0 lines) ---
  701.  
  702. <--- SIP read from TCP:10.11.1.100:45222 --->
  703. REGISTER sip:10.10.10.252:5060 SIP/2.0
  704. Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.GuaFIrLTw;rport
  705. From: <sip:[email protected]:5060>;tag=4-XZu0mXf
  706. To: sip:[email protected]:5060
  707. CSeq: 24 REGISTER
  708. Call-ID: BcqQMUJgTI
  709. Max-Forwards: 70
  710. Supported: replaces, outbound
  711. Accept: application/sdp
  712. Accept: text/plain
  713. Accept: application/vnd.gsma.rcs-ft-http+xml
  714. Contact: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  715. Expires: 3600
  716. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  717. Content-Length: 0
  718. Authorization: Digest realm="asterisk", nonce="48627d13", algorithm=MD5, username="bilal.lodhia", uri="sip:110.93.203.154", response="0f8d439984394d69e5d1be86aacc9876"
  719.  
  720.  
  721. <------------->
  722. --- (16 headers 0 lines) ---
  723. Sending to 10.11.1.100:45222 (no NAT)
  724. Sending to 10.11.1.100:45222 (no NAT)
  725.  
  726. <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
  727. SIP/2.0 401 Unauthorized
  728. Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.GuaFIrLTw;received=10.11.1.100;rport=45222
  729. From: <sip:[email protected]:5060>;tag=4-XZu0mXf
  730. To: sip:[email protected]:5060;tag=as5e6e5137
  731. Call-ID: BcqQMUJgTI
  732. CSeq: 24 REGISTER
  733. Server: Asterisk PBX 15.0.0
  734. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  735. Supported: replaces, timer
  736. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="545ad159"
  737. Content-Length: 0
  738.  
  739.  
  740. <------------>
  741. Scheduling destruction of SIP dialog 'BcqQMUJgTI' in 32000 ms (Method: REGISTER)
  742.  
  743. <--- SIP read from TCP:10.11.1.100:45222 --->
  744. REGISTER sip:10.10.10.252:5060 SIP/2.0
  745. Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.5omppCgF8;rport
  746. From: <sip:[email protected]:5060>;tag=4-XZu0mXf
  747. To: sip:[email protected]:5060
  748. CSeq: 25 REGISTER
  749. Call-ID: BcqQMUJgTI
  750. Max-Forwards: 70
  751. Supported: replaces, outbound
  752. Accept: application/sdp
  753. Accept: text/plain
  754. Accept: application/vnd.gsma.rcs-ft-http+xml
  755. Contact: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  756. Expires: 3600
  757. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  758. Content-Length: 0
  759. Authorization: Digest realm="asterisk", nonce="545ad159", algorithm=MD5, username="bilal.lodhia", uri="sip:110.93.203.154", response="455196a707c686a88e86e39b62d6de77"
  760.  
  761.  
  762. <------------->
  763. --- (16 headers 0 lines) ---
  764. Sending to 10.11.1.100:45222 (no NAT)
  765.  
  766. <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
  767. SIP/2.0 200 OK
  768. Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.5omppCgF8;received=10.11.1.100;rport=45222
  769. From: <sip:[email protected]:5060>;tag=4-XZu0mXf
  770. To: sip:[email protected]:5060;tag=as5e6e5137
  771. Call-ID: BcqQMUJgTI
  772. CSeq: 25 REGISTER
  773. Server: Asterisk PBX 15.0.0
  774. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  775. Supported: replaces, timer
  776. Expires: 3600
  777. Contact: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;expires=3600
  778. Date: Tue, 31 Oct 2017 03:08:12 GMT
  779. Content-Length: 0
  780.  
  781.  
  782. <------------>
  783. Scheduling destruction of SIP dialog 'BcqQMUJgTI' in 32000 ms (Method: REGISTER)
  784.  
  785. <--- SIP read from TCP:10.11.1.100:45222 --->
  786. SIP/2.0 200 Ok
  787. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  788. From: <sip:[email protected]:5060>;tag=as11d0abee
  789. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  790. Call-ID: [email protected]:5060
  791. CSeq: 102 INVITE
  792. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  793. Supported: replaces, outbound
  794. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  795. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  796. Content-Type: application/sdp
  797. Content-Length: 251
  798.  
  799. v=0
  800. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  801. s=Talk
  802. c=IN IP4 10.11.1.100
  803. t=0 0
  804. m=audio 7076 RTP/AVP 96 97 0 102
  805. a=rtpmap:96 opus/48000/2
  806. a=fmtp:96 useinbandfec=1
  807. a=rtpmap:97 speex/16000
  808. a=fmtp:97 vbr=on
  809. a=rtpmap:102 telephone-event/8000
  810.  
  811. <------------->
  812. --- (12 headers 11 lines) ---
  813. Transmitting (no NAT) to 10.11.1.100:45222:
  814. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  815. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3489f037
  816. Max-Forwards: 70
  817. From: <sip:[email protected]>;tag=as11d0abee
  818. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  819. Contact: <sip:[email protected]:5060;transport=tcp>
  820. Call-ID: [email protected]:5060
  821. CSeq: 102 ACK
  822. User-Agent: Asterisk PBX 15.0.0
  823. Content-Length: 0
  824.  
  825.  
  826. ---
  827.  
  828. <--- SIP read from TCP:10.11.1.100:45222 --->
  829. SIP/2.0 400 Bad request
  830. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3489f037
  831. From: <sip:[email protected]:5060>;tag=as11d0abee
  832. Call-ID: [email protected]:5060
  833. CSeq: 102 ACK
  834. Content-Length: 0
  835.  
  836.  
  837. <------------->
  838. --- (6 headers 0 lines) ---
  839.  
  840. <--- SIP read from TCP:10.11.1.100:45222 --->
  841. SIP/2.0 200 Ok
  842. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  843. From: <sip:[email protected]:5060>;tag=as11d0abee
  844. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  845. Call-ID: [email protected]:5060
  846. CSeq: 102 INVITE
  847. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  848. Supported: replaces, outbound
  849. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  850. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  851. Content-Type: application/sdp
  852. Content-Length: 251
  853.  
  854. v=0
  855. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  856. s=Talk
  857. c=IN IP4 10.11.1.100
  858. t=0 0
  859. m=audio 7076 RTP/AVP 96 97 0 102
  860. a=rtpmap:96 opus/48000/2
  861. a=fmtp:96 useinbandfec=1
  862. a=rtpmap:97 speex/16000
  863. a=fmtp:97 vbr=on
  864. a=rtpmap:102 telephone-event/8000
  865.  
  866. <------------->
  867. --- (12 headers 11 lines) ---
  868. Transmitting (no NAT) to 10.11.1.100:45222:
  869. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  870. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6e312c3f
  871. Max-Forwards: 70
  872. From: <sip:[email protected]>;tag=as11d0abee
  873. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  874. Contact: <sip:[email protected]:5060;transport=tcp>
  875. Call-ID: [email protected]:5060
  876. CSeq: 102 ACK
  877. User-Agent: Asterisk PBX 15.0.0
  878. Content-Length: 0
  879.  
  880.  
  881. ---
  882.  
  883. <--- SIP read from TCP:10.11.1.100:45222 --->
  884. SIP/2.0 400 Bad request
  885. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6e312c3f
  886. From: <sip:[email protected]:5060>;tag=as11d0abee
  887. Call-ID: [email protected]:5060
  888. CSeq: 102 ACK
  889. Content-Length: 0
  890.  
  891.  
  892. <------------->
  893. --- (6 headers 0 lines) ---
  894.  
  895. <--- SIP read from TCP:10.11.1.100:45222 --->
  896. SIP/2.0 200 Ok
  897. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  898. From: <sip:[email protected]:5060>;tag=as11d0abee
  899. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  900. Call-ID: [email protected]:5060
  901. CSeq: 102 INVITE
  902. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  903. Supported: replaces, outbound
  904. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  905. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  906. Content-Type: application/sdp
  907. Content-Length: 251
  908.  
  909. v=0
  910. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  911. s=Talk
  912. c=IN IP4 10.11.1.100
  913. t=0 0
  914. m=audio 7076 RTP/AVP 96 97 0 102
  915. a=rtpmap:96 opus/48000/2
  916. a=fmtp:96 useinbandfec=1
  917. a=rtpmap:97 speex/16000
  918. a=fmtp:97 vbr=on
  919. a=rtpmap:102 telephone-event/8000
  920.  
  921. <------------->
  922. --- (12 headers 11 lines) ---
  923. Transmitting (no NAT) to 10.11.1.100:45222:
  924. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  925. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6bab51b0
  926. Max-Forwards: 70
  927. From: <sip:[email protected]>;tag=as11d0abee
  928. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  929. Contact: <sip:[email protected]:5060;transport=tcp>
  930. Call-ID: [email protected]:5060
  931. CSeq: 102 ACK
  932. User-Agent: Asterisk PBX 15.0.0
  933. Content-Length: 0
  934.  
  935.  
  936. ---
  937.  
  938. <--- SIP read from TCP:10.11.1.100:45222 --->
  939. SIP/2.0 400 Bad request
  940. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6bab51b0
  941. From: <sip:[email protected]:5060>;tag=as11d0abee
  942. Call-ID: [email protected]:5060
  943. CSeq: 102 ACK
  944. Content-Length: 0
  945.  
  946.  
  947. <------------->
  948. --- (6 headers 0 lines) ---
  949.  
  950. <--- SIP read from TCP:10.11.1.100:45222 --->
  951. SIP/2.0 200 Ok
  952. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  953. From: <sip:[email protected]:5060>;tag=as11d0abee
  954. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  955. Call-ID: [email protected]:5060
  956. CSeq: 102 INVITE
  957. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  958. Supported: replaces, outbound
  959. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  960. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  961. Content-Type: application/sdp
  962. Content-Length: 251
  963.  
  964. v=0
  965. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  966. s=Talk
  967. c=IN IP4 10.11.1.100
  968. t=0 0
  969. m=audio 7076 RTP/AVP 96 97 0 102
  970. a=rtpmap:96 opus/48000/2
  971. a=fmtp:96 useinbandfec=1
  972. a=rtpmap:97 speex/16000
  973. a=fmtp:97 vbr=on
  974. a=rtpmap:102 telephone-event/8000
  975.  
  976. <------------->
  977. --- (12 headers 11 lines) ---
  978. Transmitting (no NAT) to 10.11.1.100:45222:
  979. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  980. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK73747bf8
  981. Max-Forwards: 70
  982. From: <sip:[email protected]>;tag=as11d0abee
  983. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  984. Contact: <sip:[email protected]:5060;transport=tcp>
  985. Call-ID: [email protected]:5060
  986. CSeq: 102 ACK
  987. User-Agent: Asterisk PBX 15.0.0
  988. Content-Length: 0
  989.  
  990.  
  991. ---
  992.  
  993. <--- SIP read from TCP:10.11.1.100:45222 --->
  994. SIP/2.0 400 Bad request
  995. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK73747bf8
  996. From: <sip:[email protected]:5060>;tag=as11d0abee
  997. Call-ID: [email protected]:5060
  998. CSeq: 102 ACK
  999. Content-Length: 0
  1000.  
  1001.  
  1002. <------------->
  1003. --- (6 headers 0 lines) ---
  1004.  
  1005. <--- SIP read from TCP:10.11.1.100:45222 --->
  1006. SIP/2.0 200 Ok
  1007. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  1008. From: <sip:[email protected]:5060>;tag=as11d0abee
  1009. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  1010. Call-ID: [email protected]:5060
  1011. CSeq: 102 INVITE
  1012. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  1013. Supported: replaces, outbound
  1014. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  1015. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  1016. Content-Type: application/sdp
  1017. Content-Length: 251
  1018.  
  1019. v=0
  1020. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  1021. s=Talk
  1022. c=IN IP4 10.11.1.100
  1023. t=0 0
  1024. m=audio 7076 RTP/AVP 96 97 0 102
  1025. a=rtpmap:96 opus/48000/2
  1026. a=fmtp:96 useinbandfec=1
  1027. a=rtpmap:97 speex/16000
  1028. a=fmtp:97 vbr=on
  1029. a=rtpmap:102 telephone-event/8000
  1030.  
  1031. <------------->
  1032. --- (12 headers 11 lines) ---
  1033. Transmitting (no NAT) to 10.11.1.100:45222:
  1034. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  1035. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK570eb103
  1036. Max-Forwards: 70
  1037. From: <sip:[email protected]>;tag=as11d0abee
  1038. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  1039. Contact: <sip:[email protected]:5060;transport=tcp>
  1040. Call-ID: [email protected]:5060
  1041. CSeq: 102 ACK
  1042. User-Agent: Asterisk PBX 15.0.0
  1043. Content-Length: 0
  1044.  
  1045.  
  1046. ---
  1047.  
  1048. <--- SIP read from TCP:10.11.1.100:45222 --->
  1049. SIP/2.0 400 Bad request
  1050. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK570eb103
  1051. From: <sip:[email protected]:5060>;tag=as11d0abee
  1052. Call-ID: [email protected]:5060
  1053. CSeq: 102 ACK
  1054. Content-Length: 0
  1055.  
  1056.  
  1057. <------------->
  1058. --- (6 headers 0 lines) ---
  1059.  
  1060. <--- SIP read from TCP:10.11.1.100:45222 --->
  1061. SIP/2.0 200 Ok
  1062. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  1063. From: <sip:[email protected]:5060>;tag=as11d0abee
  1064. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  1065. Call-ID: [email protected]:5060
  1066. CSeq: 102 INVITE
  1067. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  1068. Supported: replaces, outbound
  1069. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  1070. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  1071. Content-Type: application/sdp
  1072. Content-Length: 251
  1073.  
  1074. v=0
  1075. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  1076. s=Talk
  1077. c=IN IP4 10.11.1.100
  1078. t=0 0
  1079. m=audio 7076 RTP/AVP 96 97 0 102
  1080. a=rtpmap:96 opus/48000/2
  1081. a=fmtp:96 useinbandfec=1
  1082. a=rtpmap:97 speex/16000
  1083. a=fmtp:97 vbr=on
  1084. a=rtpmap:102 telephone-event/8000
  1085.  
  1086. <------------->
  1087. --- (12 headers 11 lines) ---
  1088. Transmitting (no NAT) to 10.11.1.100:45222:
  1089. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  1090. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK08d2b416
  1091. Max-Forwards: 70
  1092. From: <sip:[email protected]>;tag=as11d0abee
  1093. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  1094. Contact: <sip:[email protected]:5060;transport=tcp>
  1095. Call-ID: [email protected]:5060
  1096. CSeq: 102 ACK
  1097. User-Agent: Asterisk PBX 15.0.0
  1098. Content-Length: 0
  1099.  
  1100.  
  1101. ---
  1102.  
  1103. <--- SIP read from TCP:10.11.1.100:45222 --->
  1104. SIP/2.0 400 Bad request
  1105. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK08d2b416
  1106. From: <sip:[email protected]:5060>;tag=as11d0abee
  1107. Call-ID: [email protected]:5060
  1108. CSeq: 102 ACK
  1109. Content-Length: 0
  1110.  
  1111.  
  1112. <------------->
  1113. --- (6 headers 0 lines) ---
  1114.  
  1115. <--- SIP read from TCP:10.11.1.100:45222 --->
  1116. SIP/2.0 200 Ok
  1117. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
  1118. From: <sip:[email protected]:5060>;tag=as11d0abee
  1119. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
  1120. Call-ID: [email protected]:5060
  1121. CSeq: 102 INVITE
  1122. User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
  1123. Supported: replaces, outbound
  1124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  1125. Contact: <sip:[email protected]:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
  1126. Content-Type: application/sdp
  1127. Content-Length: 251
  1128.  
  1129. v=0
  1130. o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
  1131. s=Talk
  1132. c=IN IP4 10.11.1.100
  1133. t=0 0
  1134. m=audio 7076 RTP/AVP 96 97 0 102
  1135. a=rtpmap:96 opus/48000/2
  1136. a=fmtp:96 useinbandfec=1
  1137. a=rtpmap:97 speex/16000
  1138. a=fmtp:97 vbr=on
  1139. a=rtpmap:102 telephone-event/8000
  1140.  
  1141. <------------->
  1142. --- (12 headers 11 lines) ---
  1143. Transmitting (no NAT) to 10.11.1.100:45222:
  1144. ACK sip:[email protected]:45222;transport=tcp SIP/2.0
  1145. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5bca12c4
  1146. Max-Forwards: 70
  1147. From: <sip:[email protected]>;tag=as11d0abee
  1148. To: <sip:[email protected]:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
  1149. Contact: <sip:[email protected]:5060;transport=tcp>
  1150. Call-ID: [email protected]:5060
  1151. CSeq: 102 ACK
  1152. User-Agent: Asterisk PBX 15.0.0
  1153. Content-Length: 0
  1154.  
  1155.  
  1156. ---
  1157.  
  1158. <--- SIP read from TCP:10.11.1.100:45222 --->
  1159. SIP/2.0 400 Bad request
  1160. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5bca12c4
  1161. From: <sip:[email protected]:5060>;tag=as11d0abee
  1162. Call-ID: [email protected]:5060
  1163. CSeq: 102 ACK
  1164. Content-Length: 0
  1165.  
  1166.  
  1167. <------------->
  1168. --- (6 headers 0 lines) ---
  1169.  
  1170. <--- SIP read from TCP:10.11.1.100:45222 --->
  1171. BYE sip:[email protected]:5060;transport=tcp SIP/2.0
  1172. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.UFqQa6n~L;rport
  1173. From: <sip:[email protected];app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1>;tag=CPj1QSM
  1174. To: <sip:[email protected]:5060>;tag=as11d0abee
  1175. CSeq: 111 BYE
  1176. Call-ID: [email protected]:5060
  1177. Max-Forwards: 70
  1178. Content-Length: 0
  1179.  
  1180.  
  1181. <------------->
  1182. --- (8 headers 0 lines) ---
  1183. Sending to 10.11.1.100:45222 (no NAT)
  1184. Audio is at 11938
  1185. Adding codec opus to SDP
  1186. Adding non-codec 0x1 (telephone-event) to SDP
  1187. Reliably Transmitting (no NAT) to 10.11.3.58:63611:
  1188. INVITE sip:[email protected]:63611;transport=tcp SIP/2.0
  1189. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
  1190. Max-Forwards: 70
  1191. From: sip:[email protected];tag=as5f4d7d62
  1192. To: <sip:[email protected]>;tag=WGTTVGU85
  1193. Contact: <sip:[email protected]:5060;transport=tcp>
  1194. Call-ID: UYTfLSiokR
  1195. CSeq: 103 INVITE
  1196. User-Agent: Asterisk PBX 15.0.0
  1197. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1198. Supported: replaces, timer
  1199. X-asterisk-Info: SIP re-invite (External RTP bridge)
  1200. Content-Type: application/sdp
  1201. Content-Length: 244
  1202.  
  1203. v=0
  1204. o=root 1076832751 1076832753 IN IP4 10.10.10.252
  1205. s=Asterisk PBX 15.0.0
  1206. c=IN IP4 10.10.10.252
  1207. t=0 0
  1208. m=audio 11938 RTP/AVPF 96 102
  1209. a=rtpmap:96 opus/48000/2
  1210. a=rtpmap:102 telephone-event/8000
  1211. a=fmtp:102 0-16
  1212. a=maxptime:60
  1213. a=sendrecv
  1214.  
  1215. ---
  1216. Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: ACK)
  1217. Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)
  1218.  
  1219. <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
  1220. SIP/2.0 200 OK
  1221. Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.UFqQa6n~L;received=10.11.1.100;rport=45222
  1222. From: <sip:[email protected];app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1>;tag=CPj1QSM
  1223. To: <sip:[email protected]:5060>;tag=as11d0abee
  1224. Call-ID: [email protected]:5060
  1225. CSeq: 111 BYE
  1226. Server: Asterisk PBX 15.0.0
  1227. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1228. Supported: replaces, timer
  1229. Content-Length: 0
  1230.  
  1231.  
  1232. <------------>
  1233.  
  1234. <--- SIP read from TCP:10.11.3.58:63611 --->
  1235. SIP/2.0 100 Trying
  1236. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
  1237. From: <sip:[email protected]:5060>;tag=as5f4d7d62
  1238. To: <sip:[email protected]:5060>;tag=WGTTVGU85
  1239. Call-ID: UYTfLSiokR
  1240. CSeq: 103 INVITE
  1241. Content-Length: 0
  1242.  
  1243.  
  1244. <------------->
  1245. --- (7 headers 0 lines) ---
  1246.  
  1247. <--- SIP read from TCP:10.11.3.58:63611 --->
  1248. SIP/2.0 200 Ok
  1249. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
  1250. From: <sip:[email protected]:5060>;tag=as5f4d7d62
  1251. To: <sip:[email protected]:5060>;tag=WGTTVGU85
  1252. Call-ID: UYTfLSiokR
  1253. CSeq: 103 INVITE
  1254. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  1255. Supported: replaces, outbound
  1256. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  1257. Contact: <sip:[email protected]:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
  1258. Content-Type: application/sdp
  1259. Content-Length: 219
  1260.  
  1261. v=0
  1262. o=God 1019 1496 IN IP4 10.11.3.58
  1263. s=Talk
  1264. c=IN IP4 10.11.3.58
  1265. t=0 0
  1266. m=audio 7078 RTP/AVPF 96 102
  1267. a=rtpmap:96 opus/48000/2
  1268. a=fmtp:96 useinbandfec=1
  1269. a=rtpmap:102 telephone-event/8000
  1270. a=rtcp-fb:* trr-int 5000
  1271.  
  1272. <------------->
  1273. --- (12 headers 10 lines) ---
  1274. Found RTP audio format 96
  1275. Found RTP audio format 102
  1276. Found audio description format opus for ID 96
  1277. Found audio description format telephone-event for ID 102
  1278. Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
  1279. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  1280. Peer audio RTP is at port 10.11.3.58:7078
  1281. Transmitting (no NAT) to 10.11.3.58:63611:
  1282. ACK sip:[email protected]:63611;transport=tcp SIP/2.0
  1283. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK2ce0e61c;rport
  1284. Max-Forwards: 70
  1285. From: sip:[email protected];tag=as5f4d7d62
  1286. To: <sip:[email protected]>;tag=WGTTVGU85
  1287. Contact: <sip:[email protected]:5060;transport=tcp>
  1288. Call-ID: UYTfLSiokR
  1289. CSeq: 103 ACK
  1290. User-Agent: Asterisk PBX 15.0.0
  1291. Content-Length: 0
  1292.  
  1293.  
  1294. ---
  1295.  
  1296. <--- SIP read from TCP:10.11.3.58:63611 --->
  1297. BYE sip:[email protected]:5060;transport=tcp SIP/2.0
  1298. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.WY1oKxTvc;rport
  1299. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  1300. To: <sip:[email protected]:5060>;tag=as5f4d7d62
  1301. CSeq: 22 BYE
  1302. Call-ID: UYTfLSiokR
  1303. Max-Forwards: 70
  1304. User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
  1305. Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:[email protected]:5060;transport=tcp", response="795ced656f889150cd12c9d71901f47e"
  1306. Content-Length: 0
  1307.  
  1308.  
  1309. <------------->
  1310. --- (10 headers 0 lines) ---
  1311. Sending to 10.11.3.58:63611 (no NAT)
  1312. Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: BYE)
  1313.  
  1314. <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
  1315. SIP/2.0 200 OK
  1316. Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.WY1oKxTvc;received=10.11.3.58;rport=63611
  1317. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  1318. To: <sip:[email protected]:5060>;tag=as5f4d7d62
  1319. Call-ID: UYTfLSiokR
  1320. CSeq: 22 BYE
  1321. Server: Asterisk PBX 15.0.0
  1322. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1323. Supported: replaces, timer
  1324. Content-Length: 0
  1325.  
  1326.  
  1327. <------------>
  1328. Reliably Transmitting (no NAT) to 10.11.3.58:63611:
  1329. BYE sip:[email protected]:63611;transport=tcp SIP/2.0
  1330. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK348950c6;rport
  1331. Max-Forwards: 70
  1332. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  1333. To: <sip:[email protected]:5060>;tag=as5f4d7d62
  1334. Call-ID: UYTfLSiokR
  1335. CSeq: 104 BYE
  1336. User-Agent: Asterisk PBX 15.0.0
  1337. Proxy-Authorization: Digest username="God", realm="asterisk", algorithm=MD5, uri="sip:110.93.203.154", nonce="59fb78f5", response="01c565d00b73c5085afbcff1302d214a"
  1338. X-Asterisk-HangupCause: Normal Clearing
  1339. X-Asterisk-HangupCauseCode: 16
  1340. Content-Length: 0
  1341.  
  1342.  
  1343. ---
  1344.  
  1345. <--- SIP read from TCP:10.11.3.58:63611 --->
  1346. SIP/2.0 481 Call/transaction does not exist
  1347. Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK348950c6;rport
  1348. From: <sip:[email protected]:5060>;tag=WGTTVGU85
  1349. To: <sip:[email protected]:5060>;tag=as5f4d7d62
  1350. Call-ID: UYTfLSiokR
  1351. CSeq: 104 BYE
  1352. Content-Length: 0
  1353.  
  1354.  
  1355. <------------->
  1356. --- (7 headers 0 lines) ---
  1357. Really destroying SIP dialog 'BcqQMUJgTI' Method: REGISTER
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