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  1. -- Executing [104@DEFAULT:1] Dial("SIP/106-00000004", "SIP/104,30") in new stack
  2. == Using SIP RTP CoS mark 5
  3. Audio is at 14896
  4. Video is at 192.168.1.2:12062
  5. Adding codec 0x8 (alaw) to SDP
  6. Adding codec 0x4 (ulaw) to SDP
  7. Adding video codec 0x200000 (h264) to SDP
  8. Adding non-codec 0x1 (telephone-event) to SDP
  9. Reliably Transmitting (no NAT) to 78.36.193.173:59792:
  10. INVITE sip:104@78.36.193.173:59792;ob SIP/2.0
  11. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK677ce7de
  12. Max-Forwards: 70
  13. From: "user6" <sip:106@192.168.1.2:7260>;tag=as11865e8a
  14. To: <sip:104@78.36.193.173:59792;ob>
  15. Contact: <sip:106@192.168.1.2:7260>
  16. Call-ID: 205231cb1bc0889e3f1ebd2b59816bed@192.168.1.2:7260
  17. CSeq: 102 INVITE
  18. User-Agent: Asterisk PBX 1.8.10.1
  19. Date: Tue, 06 Mar 2018 14:20:25 GMT
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  21. Supported: replaces, timer
  22. Content-Type: application/sdp
  23. Content-Length: 358
  24.  
  25. v=0
  26. o=root 1978169929 1978169929 IN IP4 192.168.1.2
  27. s=Asterisk PBX 1.8.10.1
  28. c=IN IP4 192.168.1.2
  29. b=CT:384
  30. t=0 0
  31. m=audio 14896 RTP/AVP 8 0 101
  32. a=rtpmap:8 PCMA/8000
  33. a=rtpmap:0 PCMU/8000
  34. a=rtpmap:101 telephone-event/8000
  35. a=fmtp:101 0-16
  36. a=silenceSupp:off - - - -
  37. a=ptime:20
  38. a=sendrecv
  39. m=video 12062 RTP/AVP 99
  40. a=rtpmap:99 H264/90000
  41. a=sendrecv
  42.  
  43. ---
  44. -- Called SIP/104
  45.  
  46. <--- SIP read from UDP:78.36.193.173:59792 --->
  47. SIP/2.0 100 Trying
  48. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK677ce7de
  49. Call-ID: 205231cb1bc0889e3f1ebd2b59816bed@192.168.1.2:7260
  50. From: "user6" <sip:106@192.168.1.2>;tag=as11865e8a
  51. To: <sip:104@78.36.193.173;ob>
  52. CSeq: 102 INVITE
  53. Content-Length: 0
  54.  
  55. <------------->
  56. --- (7 headers 0 lines) ---
  57.  
  58. <--- SIP read from UDP:78.36.193.173:59792 --->
  59. SIP/2.0 486 Busy Here
  60. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK677ce7de
  61. Call-ID: 205231cb1bc0889e3f1ebd2b59816bed@192.168.1.2:7260
  62. From: "user6" <sip:106@192.168.1.2>;tag=as11865e8a
  63. To: <sip:104@78.36.193.173;ob>;tag=0782407eb4bd4f10bbfc3fa56d0ae021
  64. CSeq: 102 INVITE
  65. Content-Length: 0
  66.  
  67. <------------->
  68. --- (7 headers 0 lines) ---
  69. -- Got SIP response 486 "Busy Here" back from 78.36.193.173:59792
  70. Transmitting (no NAT) to 78.36.193.173:59792:
  71. ACK sip:104@78.36.193.173:59792;ob SIP/2.0
  72. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK677ce7de
  73. Max-Forwards: 70
  74. From: "user6" <sip:106@192.168.1.2:7260>;tag=as11865e8a
  75. To: <sip:104@78.36.193.173:59792;ob>;tag=0782407eb4bd4f10bbfc3fa56d0ae021
  76. Contact: <sip:106@192.168.1.2:7260>
  77. Call-ID: 205231cb1bc0889e3f1ebd2b59816bed@192.168.1.2:7260
  78. CSeq: 102 ACK
  79. User-Agent: Asterisk PBX 1.8.10.1
  80. Content-Length: 0
  81.  
  82.  
  83. ---
  84. -- SIP/104-00000005 is busy
  85. == Everyone is busy/congested at this time (1:1/0/0)
  86. -- Executing [104@DEFAULT:2] Wait("SIP/106-00000004", "5") in new stack
  87. Really destroying SIP dialog '205231cb1bc0889e3f1ebd2b59816bed@192.168.1.2:7260' Method: INVITE
  88. Retransmitting #9 (no NAT) to 78.36.193.173:7260:
  89. SIP/2.0 200 OK
  90. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.Y8LsMTNC7;received=78.36.193.173;rport=7260
  91. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=5n3C8RY5B
  92. To: sip:104@763c08941b00.sn.mynetname.net;tag=as54b40a87
  93. Call-ID: QKw5mvTA8t
  94. CSeq: 21 INVITE
  95. Server: Asterisk PBX 1.8.10.1
  96. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  97. Supported: replaces, timer
  98. Contact: <sip:104@192.168.1.2:7260>
  99. Content-Type: application/sdp
  100. Content-Length: 358
  101.  
  102. v=0
  103. o=root 1280843302 1280843302 IN IP4 192.168.1.2
  104. s=Asterisk PBX 1.8.10.1
  105. c=IN IP4 192.168.1.2
  106. b=CT:384
  107. t=0 0
  108. m=audio 17830 RTP/AVP 8 0 100
  109. a=rtpmap:8 PCMA/8000
  110. a=rtpmap:0 PCMU/8000
  111. a=rtpmap:100 telephone-event/8000
  112. a=fmtp:100 0-16
  113. a=silenceSupp:off - - - -
  114. a=ptime:20
  115. a=sendrecv
  116. m=video 15822 RTP/AVP 97
  117. a=rtpmap:97 H264/90000
  118. a=sendrecv
  119.  
  120. ---
  121. -- Executing [104@DEFAULT:3] Dial("SIP/106-00000004", "SIP/103,30") in new stack
  122. Retransmitting #10 (no NAT) to 78.36.193.173:7260:
  123. SIP/2.0 200 OK
  124. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.Y8LsMTNC7;received=78.36.193.173;rport=7260
  125. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=5n3C8RY5B
  126. To: sip:104@763c08941b00.sn.mynetname.net;tag=as54b40a87
  127. Call-ID: QKw5mvTA8t
  128. CSeq: 21 INVITE
  129. Server: Asterisk PBX 1.8.10.1
  130. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  131. Supported: replaces, timer
  132. Contact: <sip:104@192.168.1.2:7260>
  133. Content-Type: application/sdp
  134. Content-Length: 358
  135.  
  136. v=0
  137. o=root 1280843302 1280843302 IN IP4 192.168.1.2
  138. s=Asterisk PBX 1.8.10.1
  139. c=IN IP4 192.168.1.2
  140. b=CT:384
  141. t=0 0
  142. m=audio 17830 RTP/AVP 8 0 100
  143. a=rtpmap:8 PCMA/8000
  144. a=rtpmap:0 PCMU/8000
  145. a=rtpmap:100 telephone-event/8000
  146. a=fmtp:100 0-16
  147. a=silenceSupp:off - - - -
  148. a=ptime:20
  149. a=sendrecv
  150. m=video 15822 RTP/AVP 97
  151. a=rtpmap:97 H264/90000
  152. a=sendrecv
  153.  
  154. ---
  155. == Using SIP RTP CoS mark 5
  156. Audio is at 18706
  157. Video is at 192.168.1.2:18050
  158. Adding codec 0x8 (alaw) to SDP
  159. Adding codec 0x4 (ulaw) to SDP
  160. Adding video codec 0x200000 (h264) to SDP
  161. Adding non-codec 0x1 (telephone-event) to SDP
  162. Reliably Transmitting (no NAT) to 192.168.1.229:64956:
  163. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  164. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  165. Max-Forwards: 70
  166. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  167. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  168. Contact: <sip:106@192.168.1.2:7260>
  169. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  170. CSeq: 102 INVITE
  171. User-Agent: Asterisk PBX 1.8.10.1
  172. Date: Tue, 06 Mar 2018 14:20:30 GMT
  173. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  174. Supported: replaces, timer
  175. Content-Type: application/sdp
  176. Content-Length: 358
  177.  
  178. v=0
  179. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  180. s=Asterisk PBX 1.8.10.1
  181. c=IN IP4 192.168.1.2
  182. b=CT:384
  183. t=0 0
  184. m=audio 18706 RTP/AVP 8 0 101
  185. a=rtpmap:8 PCMA/8000
  186. a=rtpmap:0 PCMU/8000
  187. a=rtpmap:101 telephone-event/8000
  188. a=fmtp:101 0-16
  189. a=silenceSupp:off - - - -
  190. a=ptime:20
  191. a=sendrecv
  192. m=video 18050 RTP/AVP 99
  193. a=rtpmap:99 H264/90000
  194. a=sendrecv
  195.  
  196. ---
  197. -- Called SIP/103
  198. set_destination: Parsing <sip:106@78.36.193.173:7260;transport=udp> for address/port to send to
  199. set_destination: set destination to 78.36.193.173:7260
  200. Audio is at 17830
  201. Video is at 192.168.1.2:15822
  202. Adding codec 0x8 (alaw) to SDP
  203. Adding codec 0x4 (ulaw) to SDP
  204. Adding video codec 0x200000 (h264) to SDP
  205. Adding non-codec 0x1 (telephone-event) to SDP
  206. Reliably Transmitting (no NAT) to 78.36.193.173:7260:
  207. INVITE sip:106@78.36.193.173:7260;transport=udp SIP/2.0
  208. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK00f63da4;rport
  209. Max-Forwards: 70
  210. From: sip:104@763c08941b00.sn.mynetname.net;tag=as54b40a87
  211. To: <sip:106@763c08941b00.sn.mynetname.net>;tag=5n3C8RY5B
  212. Contact: <sip:104@192.168.1.2:7260>
  213. Call-ID: QKw5mvTA8t
  214. CSeq: 102 INVITE
  215. User-Agent: Asterisk PBX 1.8.10.1
  216. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  217. Supported: replaces, timer
  218. X-asterisk-Info: SIP re-invite (External RTP bridge)
  219. Content-Type: application/sdp
  220. Content-Length: 358
  221.  
  222. v=0
  223. o=root 1280843302 1280843303 IN IP4 192.168.1.2
  224. s=Asterisk PBX 1.8.10.1
  225. c=IN IP4 192.168.1.2
  226. b=CT:384
  227. t=0 0
  228. m=audio 17830 RTP/AVP 8 0 100
  229. a=rtpmap:8 PCMA/8000
  230. a=rtpmap:0 PCMU/8000
  231. a=rtpmap:100 telephone-event/8000
  232. a=fmtp:100 0-16
  233. a=silenceSupp:off - - - -
  234. a=ptime:20
  235. a=sendrecv
  236. m=video 15822 RTP/AVP 97
  237. a=rtpmap:97 H264/90000
  238. a=sendrecv
  239.  
  240. ---
  241. set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  242. set_destination: set destination to 78.36.193.173:59792
  243. Audio is at 16582
  244. Video is at 192.168.1.2:10222
  245. Adding codec 0x8 (alaw) to SDP
  246. Adding video codec 0x200000 (h264) to SDP
  247. Adding non-codec 0x1 (telephone-event) to SDP
  248. Retransmitting #1 (no NAT) to 192.168.1.229:64956:
  249. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  250. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  251. Max-Forwards: 70
  252. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  253. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  254. Contact: <sip:106@192.168.1.2:7260>
  255. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  256. CSeq: 102 INVITE
  257. User-Agent: Asterisk PBX 1.8.10.1
  258. Date: Tue, 06 Mar 2018 14:20:30 GMT
  259. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  260. Supported: replaces, timer
  261. Content-Type: application/sdp
  262. Content-Length: 358
  263.  
  264. v=0
  265. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  266. s=Asterisk PBX 1.8.10.1
  267. c=IN IP4 192.168.1.2
  268. b=CT:384
  269. t=0 0
  270. m=audio 18706 RTP/AVP 8 0 101
  271. a=rtpmap:8 PCMA/8000
  272. a=rtpmap:0 PCMU/8000
  273. a=rtpmap:101 telephone-event/8000
  274. a=fmtp:101 0-16
  275. a=silenceSupp:off - - - -
  276. a=ptime:20
  277. a=sendrecv
  278. m=video 18050 RTP/AVP 99
  279. a=rtpmap:99 H264/90000
  280. a=sendrecv
  281.  
  282. ---
  283. Reliably Transmitting (no NAT) to 78.36.193.173:59792:
  284. INVITE sip:104@78.36.193.173:59792;ob SIP/2.0
  285. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK65cc5283
  286. Max-Forwards: 70
  287. From: "user6" <sip:106@192.168.1.2:7260>;tag=as3cb4bf2e
  288. To: <sip:104@78.36.193.173:59792;ob>;tag=5f0e73eb88844c319eb3407b44c26996
  289. Contact: <sip:106@192.168.1.2:7260>
  290. Call-ID: 3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260
  291. CSeq: 104 INVITE
  292. User-Agent: Asterisk PBX 1.8.10.1
  293. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  294. Supported: replaces, timer
  295. X-asterisk-Info: SIP re-invite (External RTP bridge)
  296. Content-Type: application/sdp
  297. Content-Length: 332
  298.  
  299. v=0
  300. o=root 616817791 616817793 IN IP4 192.168.1.2
  301. s=Asterisk PBX 1.8.10.1
  302. c=IN IP4 192.168.1.2
  303. b=CT:384
  304. t=0 0
  305. m=audio 16582 RTP/AVP 8 101
  306. a=rtpmap:8 PCMA/8000
  307. a=rtpmap:101 telephone-event/8000
  308. a=fmtp:101 0-16
  309. a=silenceSupp:off - - - -
  310. a=ptime:20
  311. a=sendrecv
  312. m=video 10222 RTP/AVP 99
  313. a=rtpmap:99 H264/90000
  314. a=sendrecv
  315.  
  316. ---
  317. Scheduling destruction of SIP dialog '3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260' in 32000 ms (Method: INVITE)
  318.  
  319. <--- SIP read from UDP:78.36.193.173:59792 --->
  320. SIP/2.0 200 OK
  321. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK65cc5283
  322. Call-ID: 3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260
  323. From: "user6" <sip:106@192.168.1.2>;tag=as3cb4bf2e
  324. To: <sip:104@78.36.193.173;ob>;tag=5f0e73eb88844c319eb3407b44c26996
  325. CSeq: 104 INVITE
  326. Contact: <sip:104@78.36.193.173:59792;ob>
  327. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  328. Supported: replaces, 100rel, timer, norefersub
  329. Content-Type: application/sdp
  330. Content-Length: 471
  331.  
  332. v=0
  333. o=- 3729341994 3729341997 IN IP4 78.36.193.173
  334. s=pjmedia
  335. b=AS:1659
  336. t=0 0
  337. a=X-nat:0
  338. m=audio 4028 RTP/AVP 8 101
  339. c=IN IP4 78.36.193.173
  340. b=TIAS:64000
  341. a=rtcp:4029 IN IP4 78.36.193.173
  342. a=sendrecv
  343. a=rtpmap:8 PCMA/8000
  344. a=rtpmap:101 telephone-event/8000
  345. a=fmtp:101 0-16
  346. m=video 4030 RTP/AVP 99
  347. c=IN IP4 78.36.193.173
  348. b=TIAS:1500000
  349. a=rtcp:4031 IN IP4 78.36.193.173
  350. a=sendrecv
  351. a=rtpmap:99 H264/90000
  352. a=fmtp:99 profile-level-id=42000a; packetization-mode=0
  353. <------------->
  354. == Spawn extension (DEFAULT, 104, 1) exited non-zero on 'SIP/106-00000002'
  355. --- (11 headers 21 lines) ---
  356. Found RTP audio format 8
  357. Found RTP audio format 101
  358. Scheduling destruction of SIP dialog 'QKw5mvTA8t' in 32000 ms (Method: INVITE)
  359. Found audio description format PCMA for ID 8
  360. Found audio description format telephone-event for ID 101
  361. Found RTP video format 99
  362. Found video description format H264 for ID 99
  363. Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x8 (alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200008 (alaw|h264)
  364. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  365. Peer audio RTP is at port 78.36.193.173:4028
  366. Peer video RTP is at port 78.36.193.173:4030
  367. set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  368. set_destination: set destination to 78.36.193.173:59792
  369. Transmitting (no NAT) to 78.36.193.173:59792:
  370. ACK sip:104@78.36.193.173:59792;ob SIP/2.0
  371. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK262aeb5a
  372. Max-Forwards: 70
  373. From: "user6" <sip:106@192.168.1.2:7260>;tag=as3cb4bf2e
  374. To: <sip:104@78.36.193.173:59792;ob>;tag=5f0e73eb88844c319eb3407b44c26996
  375. Contact: <sip:106@192.168.1.2:7260>
  376. Call-ID: 3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260
  377. CSeq: 104 ACK
  378. User-Agent: Asterisk PBX 1.8.10.1
  379. Content-Length: 0
  380.  
  381.  
  382. ---
  383. set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  384. set_destination: set destination to 78.36.193.173:59792
  385. Reliably Transmitting (no NAT) to 78.36.193.173:59792:
  386. BYE sip:104@78.36.193.173:59792;ob SIP/2.0
  387. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK6c9b5c95
  388. Max-Forwards: 70
  389. From: "user6" <sip:106@192.168.1.2:7260>;tag=as3cb4bf2e
  390. To: <sip:104@78.36.193.173:59792;ob>;tag=5f0e73eb88844c319eb3407b44c26996
  391. Call-ID: 3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260
  392. CSeq: 105 BYE
  393. User-Agent: Asterisk PBX 1.8.10.1
  394. X-Asterisk-HangupCause: Normal Clearing
  395. X-Asterisk-HangupCauseCode: 16
  396. Content-Length: 0
  397.  
  398.  
  399. ---
  400. Scheduling destruction of SIP dialog '3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260' in 32000 ms (Method: INVITE)
  401.  
  402. <--- SIP read from UDP:78.36.193.173:7260 --->
  403. SIP/2.0 100 Trying
  404. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK00f63da4;rport
  405. From: <sip:104@763c08941b00.sn.mynetname.net>;tag=as54b40a87
  406. To: <sip:106@763c08941b00.sn.mynetname.net>;tag=5n3C8RY5B
  407. Call-ID: QKw5mvTA8t
  408. CSeq: 102 INVITE
  409.  
  410. <------------->
  411. --- (6 headers 0 lines) ---
  412.  
  413. <--- SIP read from UDP:78.36.193.173:59792 --->
  414. SIP/2.0 200 OK
  415. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK6c9b5c95
  416. Call-ID: 3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260
  417. From: "user6" <sip:106@192.168.1.2>;tag=as3cb4bf2e
  418. To: <sip:104@78.36.193.173;ob>;tag=5f0e73eb88844c319eb3407b44c26996
  419. CSeq: 105 BYE
  420. Content-Length: 0
  421.  
  422. <------------->
  423. --- (7 headers 0 lines) ---
  424. eally destroying SIP dialog '3e343d38326901237c4a4f7559e481cf@192.168.1.2:7260' Method: INVITE
  425.  
  426. <--- SIP read from UDP:78.36.193.173:7260 --->
  427. CANCEL sip:104@763c08941b00.sn.mynetname.net SIP/2.0
  428. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.6WYIVGg5k;rport
  429. Call-ID: 6021qzzwCU
  430. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=UIsFbxp0p
  431. To: sip:104@763c08941b00.sn.mynetname.net
  432. Max-Forwards: 70
  433. CSeq: 21 CANCEL
  434. User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)
  435.  
  436. <------------->
  437. --- (8 headers 0 lines) ---
  438. Sending to 78.36.193.173:7260 (no NAT)
  439.  
  440. <--- Reliably Transmitting (no NAT) to 78.36.193.173:7260 --->
  441. SIP/2.0 487 Request Terminated
  442. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.6WYIVGg5k;received=78.36.193.173;rport=7260
  443. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=UIsFbxp0p
  444. To: sip:104@763c08941b00.sn.mynetname.net;tag=as5a4d94a0
  445. Call-ID: 6021qzzwCU
  446. CSeq: 21 INVITE
  447. Server: Asterisk PBX 1.8.10.1
  448. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  449. Supported: replaces, timer
  450. Content-Length: 0
  451.  
  452.  
  453. ------------>
  454.  
  455. <--- Transmitting (no NAT) to 78.36.193.173:7260 --->
  456. SIP/2.0 200 OK
  457. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.6WYIVGg5k;received=78.36.193.173;rport=7260
  458. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=UIsFbxp0p
  459. To: sip:104@763c08941b00.sn.mynetname.net;tag=as5a4d94a0
  460. Call-ID: 6021qzzwCU
  461. CSeq: 21 CANCEL
  462. Server: Asterisk PBX 1.8.10.1
  463. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  464. Supported: replaces, timer
  465. Content-Length: 0
  466.  
  467.  
  468. <------------>
  469. Scheduling destruction of SIP dialog '6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260' in 32000 ms (Method: INVITE)
  470. == Spawn extension (DEFAULT, 104, 3) exited non-zero on 'SIP/106-00000004'
  471.  
  472. <--- SIP read from UDP:78.36.193.173:7260 --->
  473. ACK sip:104@763c08941b00.sn.mynetname.net SIP/2.0
  474. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.6WYIVGg5k;rport
  475. Call-ID: 6021qzzwCU
  476. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=UIsFbxp0p
  477. To: <sip:104@763c08941b00.sn.mynetname.net>;tag=as5a4d94a0
  478. Contact: <sip:106@78.36.193.173:7260;transport=udp>;+sip.instance="<urn:uuid:1523d154-69f6-4727-806a-296b5e1e390c>"
  479. Max-Forwards: 70
  480. CSeq: 21 ACK
  481.  
  482. <------------->
  483. --- (8 headers 0 lines) ---
  484. Really destroying SIP dialog '6021qzzwCU' Method: ACK
  485. Retransmitting #2 (no NAT) to 192.168.1.229:64956:
  486. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  487. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  488. Max-Forwards: 70
  489. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  490. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  491. Contact: <sip:106@192.168.1.2:7260>
  492. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  493. CSeq: 102 INVITE
  494. User-Agent: Asterisk PBX 1.8.10.1
  495. Date: Tue, 06 Mar 2018 14:20:30 GMT
  496. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  497. Supported: replaces, timer
  498. Content-Type: application/sdp
  499. Content-Length: 358
  500.  
  501. v=0
  502. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  503. s=Asterisk PBX 1.8.10.1
  504. c=IN IP4 192.168.1.2
  505. b=CT:384
  506. t=0 0
  507. m=audio 18706 RTP/AVP 8 0 101
  508. a=rtpmap:8 PCMA/8000
  509. a=rtpmap:0 PCMU/8000
  510. a=rtpmap:101 telephone-event/8000
  511. a=fmtp:101 0-16
  512. a=silenceSupp:off - - - -
  513. a=ptime:20
  514. a=sendrecv
  515. m=video 18050 RTP/AVP 99
  516. a=rtpmap:99 H264/90000
  517. a=sendrecv
  518.  
  519. ---
  520. Retransmitting #3 (no NAT) to 192.168.1.229:64956:
  521. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  522. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  523. Max-Forwards: 70
  524. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  525. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  526. Contact: <sip:106@192.168.1.2:7260>
  527. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  528. CSeq: 102 INVITE
  529. User-Agent: Asterisk PBX 1.8.10.1
  530. Date: Tue, 06 Mar 2018 14:20:30 GMT
  531. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  532. Supported: replaces, timer
  533. Content-Type: application/sdp
  534. Content-Length: 358
  535.  
  536. v=0
  537. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  538. s=Asterisk PBX 1.8.10.1
  539. c=IN IP4 192.168.1.2
  540. b=CT:384
  541. t=0 0
  542. m=audio 18706 RTP/AVP 8 0 101
  543. a=rtpmap:8 PCMA/8000
  544. a=rtpmap:0 PCMU/8000
  545. a=rtpmap:101 telephone-event/8000
  546. a=fmtp:101 0-16
  547. a=silenceSupp:off - - - -
  548. a=ptime:20
  549. a=sendrecv
  550. m=video 18050 RTP/AVP 99
  551. a=rtpmap:99 H264/90000
  552. a=sendrecv
  553.  
  554. ---
  555.  
  556. <--- SIP read from UDP:78.36.193.173:59792 --->
  557.  
  558. <------------->
  559.  
  560. <--- SIP read from UDP:192.168.1.3:7260 --->
  561.  
  562. <------------->
  563.  
  564. <--- SIP read from UDP:192.168.1.5:7260 --->
  565.  
  566. <------------->
  567. Retransmitting #4 (no NAT) to 192.168.1.229:64956:
  568. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  569. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  570. Max-Forwards: 70
  571. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  572. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  573. Contact: <sip:106@192.168.1.2:7260>
  574. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  575. CSeq: 102 INVITE
  576. User-Agent: Asterisk PBX 1.8.10.1
  577. Date: Tue, 06 Mar 2018 14:20:30 GMT
  578. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  579. Supported: replaces, timer
  580. Content-Type: application/sdp
  581. Content-Length: 358
  582.  
  583. v=0
  584. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  585. s=Asterisk PBX 1.8.10.1
  586. c=IN IP4 192.168.1.2
  587. b=CT:384
  588. t=0 0
  589. m=audio 18706 RTP/AVP 8 0 101
  590. a=rtpmap:8 PCMA/8000
  591. a=rtpmap:0 PCMU/8000
  592. a=rtpmap:101 telephone-event/8000
  593. a=fmtp:101 0-16
  594. a=silenceSupp:off - - - -
  595. a=ptime:20
  596. a=sendrecv
  597. m=video 18050 RTP/AVP 99
  598. a=rtpmap:99 H264/90000
  599. a=sendrecv
  600.  
  601. ---
  602. metarouter*CLI>
  603. Disconnected from Asterisk server
  604. Executing last minute cleanups
  605. root@metarouter:~# echo > /var/log/asterisk/full
  606. root@metarouter:~# asterisk -rv
  607. Asterisk 1.8.10.1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  608. Created by Mark Spencer <markster@digium.com>
  609. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  610. This is free software, with components licensed under the GNU General Public
  611. License version 2 and other licenses; you are welcome to redistribute it under
  612. certain conditions. Type 'core show license' for details.
  613. =========================================================================
  614. Connected to Asterisk 1.8.10.1 currently running on metarouter (pid = 28675)
  615. Verbosity is at least 3
  616. -- Remote UNIX connection
  617.  
  618. <--- SIP read from UDP:78.36.193.173:7260 --->
  619.  
  620.  
  621. <------------->
  622. metarouter*CLI> sip set debug off\
  623. SIP Debugging Disabled
  624. metarouter*CLI>
  625. Disconnected from Asterisk server
  626. Executing last minute cleanups
  627. root@metarouter:~# asterisk -rv
  628. Asterisk 1.8.10.1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  629. Created by Mark Spencer <markster@digium.com>
  630. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  631. This is free software, with components licensed under the GNU General Public
  632. License version 2 and other licenses; you are welcome to redistribute it under
  633. certain conditions. Type 'core show license' for details.
  634. =========================================================================
  635. Connected to Asterisk 1.8.10.1 currently running on metarouter (pid = 28675)
  636. Verbosity is at least 3
  637. metarouter*CLI> sip set debug off
  638. SIP Debugging Disabled
  639. metarouter*CLI> sip set debug off\
  640. Disconnected from Asterisk server
  641. Executing last minute cleanups
  642. root@metarouter:~#
  643. root@metarouter:~# cat /var/log/asterisk/full
  644.  
  645. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  646. <--- SIP read from UDP:78.36.193.173:7260 --->
  647. INVITE sip:104@763c08941b00.sn.mynetname.net SIP/2.0
  648. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.vc1BsfFFJ;rport
  649. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  650. To: sip:104@763c08941b00.sn.mynetname.net
  651. CSeq: 20 INVITE
  652. Call-ID: 5T9ZZw462i
  653. Max-Forwards: 70
  654. Supported: replaces, outbound
  655. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  656. Content-Type: application/sdp
  657. Content-Length: 753
  658. Contact: <sip:106@78.36.193.173:7260;transport=udp>;+sip.instance="<urn:uuid:1523d154-69f6-4727-806a-296b5e1e390c>"
  659. User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)
  660.  
  661. v=0
  662. o=106 1667 327 IN IP4 10.2.2.96
  663. s=Talk
  664. c=IN IP4 10.2.2.96
  665. t=0 0
  666. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  667. m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100
  668. a=rtpmap:96 opus/48000/2
  669. a=fmtp:96 useinbandfec=1
  670. a=rtpmap:97 speex/16000
  671. a=fmtp:97 vbr=on
  672. a=rtpmap:98 speex/8000
  673. a=fmtp:98 vbr=on
  674. a=fmtp:18 annexb=yes
  675. a=rtpmap:101 telephone-event/48000
  676. a=rtpmap:99 telephone-event/16000
  677. a=rtpmap:100 telephone-event/8000
  678. a=rtcp-fb:* ccm tmmbr
  679. m=video 9078 RTP/AVP 96 97
  680. a=rtpmap:96 VP8/90000
  681. a=rtpmap:97 H264/90000
  682. a=fmtp:97 profile-level-id=42801F
  683. a=rtcp-fb:* ccm tmmbr
  684. a=rtcp-fb:96 nack pli
  685. a=rtcp-fb:96 nack sli
  686. a=rtcp-fb:96 ack rpsi
  687. a=rtcp-fb:96 ccm fir
  688. a=rtcp-fb:97 nack pli
  689. a=rtcp-fb:97 ccm fir
  690. <------------->
  691. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: --- (13 headers 29 lines) ---
  692. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Sending to 78.36.193.173:7260 (NAT)
  693. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Using INVITE request as basis request - 5T9ZZw462i
  694. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found peer '106' for '106' from 78.36.193.173:7260
  695. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  696. <--- Reliably Transmitting (no NAT) to 78.36.193.173:7260 --->
  697. SIP/2.0 401 Unauthorized
  698. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.vc1BsfFFJ;received=78.36.193.173;rport=7260
  699. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  700. To: sip:104@763c08941b00.sn.mynetname.net;tag=as33afb673
  701. Call-ID: 5T9ZZw462i
  702. CSeq: 20 INVITE
  703. Server: Asterisk PBX 1.8.10.1
  704. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  705. Supported: replaces, timer
  706. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="395f1971"
  707. Content-Length: 0
  708.  
  709.  
  710. <------------>
  711. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Scheduling destruction of SIP dialog '5T9ZZw462i' in 32000 ms (Method: INVITE)
  712. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  713. <--- SIP read from UDP:78.36.193.173:7260 --->
  714. ACK sip:104@763c08941b00.sn.mynetname.net SIP/2.0
  715. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.vc1BsfFFJ;rport
  716. Call-ID: 5T9ZZw462i
  717. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  718. To: <sip:104@763c08941b00.sn.mynetname.net>;tag=as33afb673
  719. Contact: <sip:106@78.36.193.173:7260;transport=udp>;+sip.instance="<urn:uuid:1523d154-69f6-4727-806a-296b5e1e390c>"
  720. Max-Forwards: 70
  721. CSeq: 20 ACK
  722.  
  723. <------------->
  724. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: --- (8 headers 0 lines) ---
  725. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  726. <--- SIP read from UDP:78.36.193.173:7260 --->
  727. INVITE sip:104@763c08941b00.sn.mynetname.net SIP/2.0
  728. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;rport
  729. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  730. To: sip:104@763c08941b00.sn.mynetname.net
  731. CSeq: 21 INVITE
  732. Call-ID: 5T9ZZw462i
  733. Max-Forwards: 70
  734. Supported: replaces, outbound
  735. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  736. Content-Type: application/sdp
  737. Content-Length: 753
  738. Contact: <sip:106@78.36.193.173:7260;transport=udp>;+sip.instance="<urn:uuid:1523d154-69f6-4727-806a-296b5e1e390c>"
  739. User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)
  740. Authorization: Digest realm="asterisk", nonce="395f1971", algorithm=MD5, username="106", uri="sip:104@763c08941b00.sn.mynetname.net", response="41d5d04d8b3a254329a3b505d9e5e370"
  741.  
  742. v=0
  743. o=106 1667 327 IN IP4 10.2.2.96
  744. s=Talk
  745. c=IN IP4 10.2.2.96
  746. t=0 0
  747. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  748. m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100
  749. a=rtpmap:96 opus/48000/2
  750. a=fmtp:96 useinbandfec=1
  751. a=rtpmap:97 speex/16000
  752. a=fmtp:97 vbr=on
  753. a=rtpmap:98 speex/8000
  754. a=fmtp:98 vbr=on
  755. a=fmtp:18 annexb=yes
  756. a=rtpmap:101 telephone-event/48000
  757. a=rtpmap:99 telephone-event/16000
  758. a=rtpmap:100 telephone-event/8000
  759. a=rtcp-fb:* ccm tmmbr
  760. m=video 9078 RTP/AVP 96 97
  761. a=rtpmap:96 VP8/90000
  762. a=rtpmap:97 H264/90000
  763. a=fmtp:97 profile-level-id=42801F
  764. a=rtcp-fb:* ccm tmmbr
  765. a=rtcp-fb:96 nack pli
  766. a=rtcp-fb:96 nack sli
  767. a=rtcp-fb:96 ack rpsi
  768. a=rtcp-fb:96 ccm fir
  769. a=rtcp-fb:97 nack pli
  770. a=rtcp-fb:97 ccm fir
  771. <------------->
  772. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: --- (14 headers 29 lines) ---
  773. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Sending to 78.36.193.173:7260 (no NAT)
  774. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Using INVITE request as basis request - 5T9ZZw462i
  775. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found peer '106' for '106' from 78.36.193.173:7260
  776. [Mar 6 14:20:43] VERBOSE[28686] netsock2.c: == Using SIP RTP CoS mark 5
  777. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 96
  778. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 97
  779. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 98
  780. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 0
  781. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 8
  782. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 18
  783. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 101
  784. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 99
  785. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP audio format 100
  786. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found unknown media description format opus for ID 96
  787. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found audio description format speex for ID 97
  788. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found audio description format speex for ID 98
  789. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found unknown media description format telephone-event for ID 101
  790. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found unknown media description format telephone-event for ID 99
  791. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found audio description format telephone-event for ID 100
  792. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP video format 96
  793. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found RTP video format 97
  794. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Found video description format H264 for ID 97
  795. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x20000030c (ulaw|alaw|g729|speex|speex16)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x20000c (ulaw|alaw|h264)
  796. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  797. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Peer audio RTP is at port 10.2.2.96:7076
  798. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Peer video RTP is at port 10.2.2.96:9078
  799. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: Looking for 104 in DEFAULT (domain 763c08941b00.sn.mynetname.net)
  800. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: list_route: hop: <sip:106@78.36.193.173:7260;transport=udp>
  801. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  802. <--- Transmitting (no NAT) to 78.36.193.173:7260 --->
  803. SIP/2.0 100 Trying
  804. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  805. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  806. To: sip:104@763c08941b00.sn.mynetname.net
  807. Call-ID: 5T9ZZw462i
  808. CSeq: 21 INVITE
  809. Server: Asterisk PBX 1.8.10.1
  810. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  811. Supported: replaces, timer
  812. Contact: <sip:104@192.168.1.2:7260>
  813. Content-Length: 0
  814.  
  815.  
  816. <------------>
  817. [Mar 6 14:20:43] VERBOSE[28714] pbx.c: -- Executing [104@DEFAULT:1] Dial("SIP/106-00000007", "SIP/104,30") in new stack
  818. [Mar 6 14:20:43] VERBOSE[28714] netsock2.c: == Using SIP RTP CoS mark 5
  819. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Audio is at 17522
  820. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Video is at 192.168.1.2:19852
  821. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  822. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  823. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
  824. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  825. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c: Reliably Transmitting (no NAT) to 78.36.193.173:59792:
  826. INVITE sip:104@78.36.193.173:59792;ob SIP/2.0
  827. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK562c9c97
  828. Max-Forwards: 70
  829. From: "user6" <sip:106@192.168.1.2:7260>;tag=as737e6bda
  830. To: <sip:104@78.36.193.173:59792;ob>
  831. Contact: <sip:106@192.168.1.2:7260>
  832. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  833. CSeq: 102 INVITE
  834. User-Agent: Asterisk PBX 1.8.10.1
  835. Date: Tue, 06 Mar 2018 14:20:43 GMT
  836. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  837. Supported: replaces, timer
  838. Content-Type: application/sdp
  839. Content-Length: 356
  840.  
  841. v=0
  842. o=root 817342723 817342723 IN IP4 192.168.1.2
  843. s=Asterisk PBX 1.8.10.1
  844. c=IN IP4 192.168.1.2
  845. b=CT:384
  846. t=0 0
  847. m=audio 17522 RTP/AVP 8 0 101
  848. a=rtpmap:8 PCMA/8000
  849. a=rtpmap:0 PCMU/8000
  850. a=rtpmap:101 telephone-event/8000
  851. a=fmtp:101 0-16
  852. a=silenceSupp:off - - - -
  853. a=ptime:20
  854. a=sendrecv
  855. m=video 19852 RTP/AVP 99
  856. a=rtpmap:99 H264/90000
  857. a=sendrecv
  858.  
  859. ---
  860. [Mar 6 14:20:43] VERBOSE[28714] app_dial.c: -- Called SIP/104
  861. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  862. <--- SIP read from UDP:78.36.193.173:59792 --->
  863. SIP/2.0 100 Trying
  864. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK562c9c97
  865. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  866. From: "user6" <sip:106@192.168.1.2>;tag=as737e6bda
  867. To: <sip:104@78.36.193.173;ob>
  868. CSeq: 102 INVITE
  869. Content-Length: 0
  870.  
  871. <------------->
  872. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: --- (7 headers 0 lines) ---
  873. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c:
  874. <--- SIP read from UDP:78.36.193.173:59792 --->
  875. SIP/2.0 180 Ringing
  876. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK562c9c97
  877. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  878. From: "user6" <sip:106@192.168.1.2>;tag=as737e6bda
  879. To: <sip:104@78.36.193.173;ob>;tag=d3bf91bd355249e69e143a16223080f2
  880. CSeq: 102 INVITE
  881. Contact: <sip:104@78.36.193.173:59792;ob>
  882. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  883. Content-Length: 0
  884.  
  885. <------------->
  886. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: --- (9 headers 0 lines) ---
  887. [Mar 6 14:20:43] VERBOSE[28686] chan_sip.c: list_route: hop: <sip:104@78.36.193.173:59792;ob>
  888. [Mar 6 14:20:43] VERBOSE[28714] app_dial.c: -- SIP/104-00000008 is ringing
  889. [Mar 6 14:20:43] VERBOSE[28714] chan_sip.c:
  890. <--- Transmitting (no NAT) to 78.36.193.173:7260 --->
  891. SIP/2.0 180 Ringing
  892. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  893. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  894. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  895. Call-ID: 5T9ZZw462i
  896. CSeq: 21 INVITE
  897. Server: Asterisk PBX 1.8.10.1
  898. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  899. Supported: replaces, timer
  900. Contact: <sip:104@192.168.1.2:7260>
  901. Content-Length: 0
  902.  
  903.  
  904. <------------>
  905. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.1.229:64956:
  906. INVITE sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07 SIP/2.0
  907. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK1527aa8d
  908. Max-Forwards: 70
  909. From: "user6" <sip:106@192.168.1.2:7260>;tag=as62882186
  910. To: <sip:103@192.168.1.229:64956;rinstance=8dc661aabed97c07>
  911. Contact: <sip:106@192.168.1.2:7260>
  912. Call-ID: 6c6cb44a23c3a2d1607861ff312b30a3@192.168.1.2:7260
  913. CSeq: 102 INVITE
  914. User-Agent: Asterisk PBX 1.8.10.1
  915. Date: Tue, 06 Mar 2018 14:20:30 GMT
  916. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  917. Supported: replaces, timer
  918. Content-Type: application/sdp
  919. Content-Length: 358
  920.  
  921. v=0
  922. o=root 1355443706 1355443706 IN IP4 192.168.1.2
  923. s=Asterisk PBX 1.8.10.1
  924. c=IN IP4 192.168.1.2
  925. b=CT:384
  926. t=0 0
  927. m=audio 18706 RTP/AVP 8 0 101
  928. a=rtpmap:8 PCMA/8000
  929. a=rtpmap:0 PCMU/8000
  930. a=rtpmap:101 telephone-event/8000
  931. a=fmtp:101 0-16
  932. a=silenceSupp:off - - - -
  933. a=ptime:20
  934. a=sendrecv
  935. m=video 18050 RTP/AVP 99
  936. a=rtpmap:99 H264/90000
  937. a=sendrecv
  938.  
  939. ---
  940. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c:
  941. <--- SIP read from UDP:78.36.193.173:59792 --->
  942. SIP/2.0 200 OK
  943. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK562c9c97
  944. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  945. From: "user6" <sip:106@192.168.1.2>;tag=as737e6bda
  946. To: <sip:104@78.36.193.173;ob>;tag=d3bf91bd355249e69e143a16223080f2
  947. CSeq: 102 INVITE
  948. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  949. Contact: <sip:104@78.36.193.173:59792;ob>
  950. Supported: replaces, 100rel, timer, norefersub
  951. Content-Type: application/sdp
  952. Content-Length: 471
  953.  
  954. v=0
  955. o=- 3729342041 3729342042 IN IP4 78.36.193.173
  956. s=pjmedia
  957. b=AS:1659
  958. t=0 0
  959. a=X-nat:0
  960. m=audio 4036 RTP/AVP 8 101
  961. c=IN IP4 78.36.193.173
  962. b=TIAS:64000
  963. a=rtcp:4037 IN IP4 78.36.193.173
  964. a=sendrecv
  965. a=rtpmap:8 PCMA/8000
  966. a=rtpmap:101 telephone-event/8000
  967. a=fmtp:101 0-16
  968. m=video 4038 RTP/AVP 99
  969. c=IN IP4 78.36.193.173
  970. b=TIAS:1500000
  971. a=rtcp:4039 IN IP4 78.36.193.173
  972. a=sendrecv
  973. a=rtpmap:99 H264/90000
  974. a=fmtp:99 profile-level-id=42000a; packetization-mode=0
  975. <------------->
  976. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: --- (11 headers 21 lines) ---
  977. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP audio format 8
  978. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP audio format 101
  979. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found audio description format PCMA for ID 8
  980. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found audio description format telephone-event for ID 101
  981. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP video format 99
  982. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found video description format H264 for ID 99
  983. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x8 (alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200008 (alaw|h264)
  984. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  985. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Peer audio RTP is at port 78.36.193.173:4036
  986. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Peer video RTP is at port 78.36.193.173:4038
  987. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: list_route: hop: <sip:104@78.36.193.173:59792;ob>
  988. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  989. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: set_destination: set destination to 78.36.193.173:59792
  990. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Transmitting (no NAT) to 78.36.193.173:59792:
  991. ACK sip:104@78.36.193.173:59792;ob SIP/2.0
  992. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK5f5b2004
  993. Max-Forwards: 70
  994. From: "user6" <sip:106@192.168.1.2:7260>;tag=as737e6bda
  995. To: <sip:104@78.36.193.173:59792;ob>;tag=d3bf91bd355249e69e143a16223080f2
  996. Contact: <sip:106@192.168.1.2:7260>
  997. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  998. CSeq: 102 ACK
  999. User-Agent: Asterisk PBX 1.8.10.1
  1000. Content-Length: 0
  1001.  
  1002.  
  1003. ---
  1004. [Mar 6 14:20:46] VERBOSE[28714] app_dial.c: -- SIP/104-00000008 answered SIP/106-00000007
  1005. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Audio is at 14544
  1006. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Video is at 192.168.1.2:17376
  1007. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  1008. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  1009. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
  1010. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  1011. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c:
  1012. <--- Reliably Transmitting (no NAT) to 78.36.193.173:7260 --->
  1013. SIP/2.0 200 OK
  1014. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  1015. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  1016. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  1017. Call-ID: 5T9ZZw462i
  1018. CSeq: 21 INVITE
  1019. Server: Asterisk PBX 1.8.10.1
  1020. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1021. Supported: replaces, timer
  1022. Contact: <sip:104@192.168.1.2:7260>
  1023. Content-Type: application/sdp
  1024. Content-Length: 356
  1025.  
  1026. v=0
  1027. o=root 472467005 472467005 IN IP4 192.168.1.2
  1028. s=Asterisk PBX 1.8.10.1
  1029. c=IN IP4 192.168.1.2
  1030. b=CT:384
  1031. t=0 0
  1032. m=audio 14544 RTP/AVP 8 0 100
  1033. a=rtpmap:8 PCMA/8000
  1034. a=rtpmap:0 PCMU/8000
  1035. a=rtpmap:100 telephone-event/8000
  1036. a=fmtp:100 0-16
  1037. a=silenceSupp:off - - - -
  1038. a=ptime:20
  1039. a=sendrecv
  1040. m=video 17376 RTP/AVP 97
  1041. a=rtpmap:97 H264/90000
  1042. a=sendrecv
  1043.  
  1044. <------------>
  1045. [Mar 6 14:20:46] VERBOSE[28714] rtp_engine.c: -- Remotely bridging SIP/106-00000007 and SIP/104-00000008
  1046. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  1047. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: set_destination: set destination to 78.36.193.173:59792
  1048. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Audio is at 17522
  1049. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Video is at 10.2.2.96:9078
  1050. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  1051. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
  1052. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  1053. [Mar 6 14:20:46] VERBOSE[28714] chan_sip.c: Reliably Transmitting (no NAT) to 78.36.193.173:59792:
  1054. INVITE sip:104@78.36.193.173:59792;ob SIP/2.0
  1055. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK3584cb19
  1056. Max-Forwards: 70
  1057. From: "user6" <sip:106@192.168.1.2:7260>;tag=as737e6bda
  1058. To: <sip:104@78.36.193.173:59792;ob>;tag=d3bf91bd355249e69e143a16223080f2
  1059. Contact: <sip:106@192.168.1.2:7260>
  1060. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  1061. CSeq: 103 INVITE
  1062. User-Agent: Asterisk PBX 1.8.10.1
  1063. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1064. Supported: replaces, timer
  1065. X-asterisk-Info: SIP re-invite (External RTP bridge)
  1066. Content-Type: application/sdp
  1067. Content-Length: 326
  1068.  
  1069. v=0
  1070. o=root 817342723 817342724 IN IP4 10.2.2.96
  1071. s=Asterisk PBX 1.8.10.1
  1072. c=IN IP4 10.2.2.96
  1073. b=CT:384
  1074. t=0 0
  1075. m=audio 7076 RTP/AVP 8 101
  1076. a=rtpmap:8 PCMA/8000
  1077. a=rtpmap:101 telephone-event/8000
  1078. a=fmtp:101 0-16
  1079. a=silenceSupp:off - - - -
  1080. a=ptime:20
  1081. a=sendrecv
  1082. m=video 9078 RTP/AVP 99
  1083. a=rtpmap:99 H264/90000
  1084. a=sendrecv
  1085.  
  1086. ---
  1087. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c:
  1088. <--- SIP read from UDP:78.36.193.173:59792 --->
  1089. SIP/2.0 200 OK
  1090. Via: SIP/2.0/UDP 192.168.1.2:7260;received=5.11.70.35;branch=z9hG4bK3584cb19
  1091. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  1092. From: "user6" <sip:106@192.168.1.2>;tag=as737e6bda
  1093. To: <sip:104@78.36.193.173;ob>;tag=d3bf91bd355249e69e143a16223080f2
  1094. CSeq: 103 INVITE
  1095. Contact: <sip:104@78.36.193.173:59792;ob>
  1096. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  1097. Supported: replaces, 100rel, timer, norefersub
  1098. Content-Type: application/sdp
  1099. Content-Length: 471
  1100.  
  1101. v=0
  1102. o=- 3729342041 3729342043 IN IP4 78.36.193.173
  1103. s=pjmedia
  1104. b=AS:1659
  1105. t=0 0
  1106. a=X-nat:0
  1107. m=audio 4036 RTP/AVP 8 101
  1108. c=IN IP4 78.36.193.173
  1109. b=TIAS:64000
  1110. a=rtcp:4037 IN IP4 78.36.193.173
  1111. a=sendrecv
  1112. a=rtpmap:8 PCMA/8000
  1113. a=rtpmap:101 telephone-event/8000
  1114. a=fmtp:101 0-16
  1115. m=video 4038 RTP/AVP 99
  1116. c=IN IP4 78.36.193.173
  1117. b=TIAS:1500000
  1118. a=rtcp:4039 IN IP4 78.36.193.173
  1119. a=sendrecv
  1120. a=rtpmap:99 H264/90000
  1121. a=fmtp:99 profile-level-id=42000a; packetization-mode=0
  1122. <------------->
  1123. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: --- (11 headers 21 lines) ---
  1124. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP audio format 8
  1125. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP audio format 101
  1126. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found audio description format PCMA for ID 8
  1127. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found audio description format telephone-event for ID 101
  1128. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found RTP video format 99
  1129. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Found video description format H264 for ID 99
  1130. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x8 (alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200008 (alaw|h264)
  1131. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  1132. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Peer audio RTP is at port 78.36.193.173:4036
  1133. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Peer video RTP is at port 78.36.193.173:4038
  1134. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: set_destination: Parsing <sip:104@78.36.193.173:59792;ob> for address/port to send to
  1135. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: set_destination: set destination to 78.36.193.173:59792
  1136. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Transmitting (no NAT) to 78.36.193.173:59792:
  1137. ACK sip:104@78.36.193.173:59792;ob SIP/2.0
  1138. Via: SIP/2.0/UDP 192.168.1.2:7260;branch=z9hG4bK76ba9544
  1139. Max-Forwards: 70
  1140. From: "user6" <sip:106@192.168.1.2:7260>;tag=as737e6bda
  1141. To: <sip:104@78.36.193.173:59792;ob>;tag=d3bf91bd355249e69e143a16223080f2
  1142. Contact: <sip:106@192.168.1.2:7260>
  1143. Call-ID: 08328a6d6b2127ef7ca786793a155a37@192.168.1.2:7260
  1144. CSeq: 103 ACK
  1145. User-Agent: Asterisk PBX 1.8.10.1
  1146. Content-Length: 0
  1147.  
  1148.  
  1149. ---
  1150. [Mar 6 14:20:46] VERBOSE[28686] chan_sip.c: Retransmitting #1 (no NAT) to 78.36.193.173:7260:
  1151. SIP/2.0 200 OK
  1152. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  1153. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  1154. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  1155. Call-ID: 5T9ZZw462i
  1156. CSeq: 21 INVITE
  1157. Server: Asterisk PBX 1.8.10.1
  1158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1159. Supported: replaces, timer
  1160. Contact: <sip:104@192.168.1.2:7260>
  1161. Content-Type: application/sdp
  1162. Content-Length: 356
  1163.  
  1164. v=0
  1165. o=root 472467005 472467005 IN IP4 192.168.1.2
  1166. s=Asterisk PBX 1.8.10.1
  1167. c=IN IP4 192.168.1.2
  1168. b=CT:384
  1169. t=0 0
  1170. m=audio 14544 RTP/AVP 8 0 100
  1171. a=rtpmap:8 PCMA/8000
  1172. a=rtpmap:0 PCMU/8000
  1173. a=rtpmap:100 telephone-event/8000
  1174. a=fmtp:100 0-16
  1175. a=silenceSupp:off - - - -
  1176. a=ptime:20
  1177. a=sendrecv
  1178. m=video 17376 RTP/AVP 97
  1179. a=rtpmap:97 H264/90000
  1180. a=sendrecv
  1181.  
  1182. ---
  1183. [Mar 6 14:20:47] VERBOSE[28686] chan_sip.c: Retransmitting #2 (no NAT) to 78.36.193.173:7260:
  1184. SIP/2.0 200 OK
  1185. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  1186. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  1187. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  1188. Call-ID: 5T9ZZw462i
  1189. CSeq: 21 INVITE
  1190. Server: Asterisk PBX 1.8.10.1
  1191. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1192. Supported: replaces, timer
  1193. Contact: <sip:104@192.168.1.2:7260>
  1194. Content-Type: application/sdp
  1195. Content-Length: 356
  1196.  
  1197. v=0
  1198. o=root 472467005 472467005 IN IP4 192.168.1.2
  1199. s=Asterisk PBX 1.8.10.1
  1200. c=IN IP4 192.168.1.2
  1201. b=CT:384
  1202. t=0 0
  1203. m=audio 14544 RTP/AVP 8 0 100
  1204. a=rtpmap:8 PCMA/8000
  1205. a=rtpmap:0 PCMU/8000
  1206. a=rtpmap:100 telephone-event/8000
  1207. a=fmtp:100 0-16
  1208. a=silenceSupp:off - - - -
  1209. a=ptime:20
  1210. a=sendrecv
  1211. m=video 17376 RTP/AVP 97
  1212. a=rtpmap:97 H264/90000
  1213. a=sendrecv
  1214.  
  1215. ---
  1216. [Mar 6 14:20:49] VERBOSE[28686] chan_sip.c:
  1217. <--- SIP read from UDP:78.36.193.173:59792 --->
  1218.  
  1219. <------------->
  1220. [Mar 6 14:20:49] VERBOSE[28686] chan_sip.c: Retransmitting #3 (no NAT) to 78.36.193.173:7260:
  1221. SIP/2.0 200 OK
  1222. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  1223. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  1224. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  1225. Call-ID: 5T9ZZw462i
  1226. CSeq: 21 INVITE
  1227. Server: Asterisk PBX 1.8.10.1
  1228. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1229. Supported: replaces, timer
  1230. Contact: <sip:104@192.168.1.2:7260>
  1231. Content-Type: application/sdp
  1232. Content-Length: 356
  1233.  
  1234. v=0
  1235. o=root 472467005 472467005 IN IP4 192.168.1.2
  1236. s=Asterisk PBX 1.8.10.1
  1237. c=IN IP4 192.168.1.2
  1238. b=CT:384
  1239. t=0 0
  1240. m=audio 14544 RTP/AVP 8 0 100
  1241. a=rtpmap:8 PCMA/8000
  1242. a=rtpmap:0 PCMU/8000
  1243. a=rtpmap:100 telephone-event/8000
  1244. a=fmtp:100 0-16
  1245. a=silenceSupp:off - - - -
  1246. a=ptime:20
  1247. a=sendrecv
  1248. m=video 17376 RTP/AVP 97
  1249. a=rtpmap:97 H264/90000
  1250. a=sendrecv
  1251.  
  1252. ---
  1253. [Mar 6 14:20:51] VERBOSE[28686] chan_sip.c:
  1254. <--- SIP read from UDP:192.168.1.3:7260 --->
  1255.  
  1256. <------------->
  1257. [Mar 6 14:20:53] VERBOSE[28686] chan_sip.c:
  1258. <--- SIP read from UDP:192.168.1.5:7260 --->
  1259.  
  1260. <------------->
  1261. [Mar 6 14:20:53] VERBOSE[28686] chan_sip.c: Retransmitting #4 (no NAT) to 78.36.193.173:7260:
  1262. SIP/2.0 200 OK
  1263. Via: SIP/2.0/UDP 10.2.2.96:7260;branch=z9hG4bK.JoodWhn77;received=78.36.193.173;rport=7260
  1264. From: <sip:106@763c08941b00.sn.mynetname.net>;tag=x-5uG7dFq
  1265. To: sip:104@763c08941b00.sn.mynetname.net;tag=as10705c9b
  1266. Call-ID: 5T9ZZw462i
  1267. CSeq: 21 INVITE
  1268. Server: Asterisk PBX 1.8.10.1
  1269. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1270. Supported: replaces, timer
  1271. Contact: <sip:104@192.168.1.2:7260>
  1272. Content-Type: application/sdp
  1273. Content-Length: 356
  1274.  
  1275. v=0
  1276. o=root 472467005 472467005 IN IP4 192.168.1.2
  1277. s=Asterisk PBX 1.8.10.1
  1278. c=IN IP4 192.168.1.2
  1279. b=CT:384
  1280. t=0 0
  1281. m=audio 14544 RTP/AVP 8 0 100
  1282. a=rtpmap:8 PCMA/8000
  1283. a=rtpmap:0 PCMU/8000
  1284. a=rtpmap:100 telephone-event/8000
  1285. a=fmtp:100 0-16
  1286. a=silenceSupp:off - - - -
  1287. a=ptime:20
  1288. a=sendrecv
  1289. m=video 17376 RTP/AVP 97
  1290. a=rtpmap:97 H264/90000
  1291. a=sendrecv
  1292.  
  1293. ---
  1294. [Mar 6 14:20:54] VERBOSE[28677] asterisk.c: -- Remote UNIX connection
  1295. [Mar 6 14:20:54] VERBOSE[28686] chan_sip.c:
  1296. <--- SIP read from UDP:78.36.193.173:7260 --->
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