Strict Rtp Learning Complete Locking On Source Address

Posted on by admin
  1. Dropping Due To Strict Rtp Protection Will Switch To It In 3 Packets
Locking

Dropping Due To Strict Rtp Protection Will Switch To It In 3 Packets

  1. # Server A #
  2. <--- SIP read from UDP:185.53.91.27:5070 --->
  3. To: 22011972592315281<sip:22011972592315281@45.76.62.92>
  4. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;rport
  5. CSeq: 1 INVITE
  6. Max-Forwards: 70
  7. User-Agent: sipcli/v1.8
  8. Content-Length: 279
  9. v=0
  10. o=sipcli-Session 403739968 541045760 IN IP4 185.53.91.27
  11. c=IN IP4 185.53.91.27
  12. m=audio 5072 RTP/AVP 18 0 8 101
  13. a=rtpmap:18 G729/8000
  14. a=rtpmap:8 PCMA/8000
  15. a=ptime:20
  16. <------------->
  17. Sending to 185.53.91.27:5070 (no NAT)
  18. Using INVITE request as basis request - 1302bfada3c0539b1c4bd36b57286fc8
  19. No matching peer for '1005' from '185.53.91.27:5070'
  20. Found RTP audio format 18
  21. Found RTP audio format 8
  22. Found audio description format G729 for ID 18
  23. Found audio description format PCMA for ID 8
  24. Found audio description format telephone-event for ID 101
  25. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw alaw g729)/video=(nothing)/text=(nothing), combined - (ulaw alaw)
  26. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  27. > 0x7f77f402c050 -- Strict RTP learning after remote address set to: 185.53.91.27:5072
  28. Looking for 22011972592315281 in public (domain 45.76.62.92)
  29. <--- Reliably Transmitting (no NAT) to 185.53.91.27:5070 --->
  30. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  31. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  32. CSeq: 1 INVITE
  33. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  34. Content-Length: 0
  35. [Jan 12 22:16:07] NOTICE[3029][C-0000009b]: chan_sip.c:26687 handle_request_invite: Call from ' (185.53.91.27:5070) to extension '22011972592315281' rejected because extension not found in context 'public'.
  36. Scheduling destruction of SIP dialog '1302bfada3c0539b1c4bd36b57286fc8' in 32000 ms (Method: INVITE)
  37. SIP/2.0 404 Not Found
  38. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  39. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  40. CSeq: 1 INVITE
  41. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  42. Content-Length: 0
  43. Retransmitting #2 (no NAT) to 185.53.91.27:5070:
  44. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  45. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  46. CSeq: 1 INVITE
  47. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  48. Content-Length: 0
  49. INVITE sip:100@45.76.62.92:5060;transport=UDP SIP/2.0
  50. Via: SIP/2.0/UDP 71.200.12.159:61092;branch=z9hG4bK-524287-1---65a01c9f9b70d19a;rport
  51. Contact: <sip:14103101234@71.200.12.159:61092;transport=UDP>
  52. From: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  53. CSeq: 1 INVITE
  54. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  55. User-Agent: Z 5.2.19 rv2.8.99
  56. Content-Length: 602
  57. v=0
  58. s=Z
  59. t=0 0
  60. m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
  61. a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
  62. a=rtpmap:97 iLBC/8000
  63. a=rtpmap:110 speex/8000
  64. a=rtpmap:98 telephone-event/48000
  65. a=rtpmap:101 telephone-event/8000
  66. a=rtpmap:100 telephone-event/16000
  67. a=rtpmap:99 telephone-event/32000
  68. a=rtpmap:102 G726-32/8000
  69. <------------->
  70. Sending to 71.200.12.159:61092 (no NAT)
  71. Using INVITE request as basis request - chcjTcZm04RsObIGC7dLVg..
  72. No matching peer for '14103101234' from '71.200.12.159:61092'
  73. Found RTP audio format 106
  74. Found RTP audio format 3
  75. Found RTP audio format 0
  76. Found RTP audio format 97
  77. Found RTP audio format 112
  78. Found RTP audio format 101
  79. Found RTP audio format 99
  80. Found audio description format opus for ID 106
  81. Found audio description format iLBC for ID 97
  82. Found audio description format speex for ID 112
  83. Found unknown media description format telephone-event for ID 98
  84. Found audio description format telephone-event for ID 101
  85. Found unknown media description format telephone-event for ID 100
  86. Found unknown media description format telephone-event for ID 99
  87. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw gsm alaw g722 ilbc g726 opus speex speex16 speex32)/video=(nothing)/text=(nothing), combined - (ulaw alaw gsm)
  88. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  89. > 0x7f77f401f360 -- Strict RTP learning after remote address set to: 71.200.12.159:8000
  90. Looking for 100 in public (domain 45.76.62.92)
  91. sip_route_dump: route/path hop: <sip:14103101234@71.200.12.159:61092;transport=UDP>
  92. <--- Transmitting (no NAT) to 71.200.12.159:61092 --->
  93. Via: SIP/2.0/UDP 71.200.12.159:61092;branch=z9hG4bK-524287-1---65a01c9f9b70d19a;received=71.200.12.159;rport=61092
  94. From: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  95. Call-ID: chcjTcZm04RsObIGC7dLVg..
  96. Server: Asterisk PBX 16.1.1
  97. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  98. Contact: <sip:100@45.76.62.92:5060>
  99. <------------>
  100. -- Executing [100@public:1] Playback('SIP/71.200.12.159-0000003c', 'agent-alreadyon') in new stack
  101. Adding codec ulaw to SDP
  102. Adding codec gsm to SDP
  103. <--- Reliably Transmitting (no NAT) to 71.200.12.159:61092 --->
  104. Via: SIP/2.0/UDP 71.200.12.159:61092;branch=z9hG4bK-524287-1---65a01c9f9b70d19a;received=71.200.12.159;rport=61092
  105. From: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  106. To: <sip:100@45.76.62.92:5060;transport=UDP>;tag=as79a46b5c
  107. CSeq: 1 INVITE
  108. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  109. Contact: <sip:100@45.76.62.92:5060>
  110. Content-Length: 284
  111. v=0
  112. s=Asterisk PBX 16.1.1
  113. t=0 0
  114. a=rtpmap:0 PCMU/8000
  115. a=rtpmap:3 GSM/8000
  116. a=fmtp:101 0-16
  117. a=sendrecv
  118. <------------>
  119. > 0x7f77f401f360 -- Strict RTP switching to RTP target address 71.200.12.159:8000 as source
  120. <--- SIP read from UDP:71.200.12.159:61092 --->
  121. Via: SIP/2.0/UDP 71.200.12.159:61092;branch=z9hG4bK-524287-1---c9dc8cf10bfacf0b;rport
  122. Contact: <sip:14103101234@71.200.12.159:61092;transport=UDP>
  123. To: <sip:100@45.76.62.92:5060;transport=UDP>;tag=as79a46b5c
  124. From: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  125. CSeq: 1 ACK
  126. Content-Length: 0
  127. <------------->
  128. -- <SIP/71.200.12.159-0000003c> Playing 'agent-alreadyon.gsm' (language 'en')
  129. SIP/2.0 404 Not Found
  130. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  131. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  132. CSeq: 1 INVITE
  133. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  134. Content-Length: 0
  135. Retransmitting #4 (no NAT) to 185.53.91.27:5070:
  136. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  137. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  138. CSeq: 1 INVITE
  139. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  140. Content-Length: 0
  141. > 0x7f77f401f360 -- Strict RTP learning complete - Locking on source address 71.200.12.159:8000
  142. -- Executing [100@public:2] Playback('SIP/71.200.12.159-0000003c', 'beep') in new stack
  143. -- <SIP/71.200.12.159-0000003c> Playing 'beep.gsm' (language 'en')
  144. -- Executing [100@public:3] Dial('SIP/71.200.12.159-0000003c', 'SIP/100@server-b,Th') in new stack
  145. Audio is at 17804
  146. Adding codec alaw to SDP
  147. Adding non-codec 0x1 (telephone-event) to SDP
  148. Reliably Transmitting (no NAT) to 104.156.255.135:5060:
  149. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a
  150. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  151. Contact: <sip:14103101234@45.76.62.92:5060>
  152. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  153. User-Agent: Asterisk PBX 16.1.1
  154. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  155. Content-Type: application/sdp
  156. o=root 407118028 407118028 IN IP4 45.76.62.92
  157. c=IN IP4 45.76.62.92
  158. m=audio 17804 RTP/AVP 0 8 3 101
  159. a=rtpmap:8 PCMA/8000
  160. a=rtpmap:101 telephone-event/8000
  161. a=maxptime:150
  162. -- Called SIP/100@server-b
  163. <--- SIP read from UDP:104.156.255.135:5060 --->
  164. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a;received=45.76.62.92
  165. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  166. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  167. Server: Asterisk PBX 16.1.1
  168. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  169. Session-Expires: 1800;refresher=uas
  170. Content-Length: 0
  171. <------------->
  172. SIP/2.0 200 OK
  173. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a;received=45.76.62.92
  174. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  175. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  176. Server: Asterisk PBX 16.1.1
  177. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  178. Session-Expires: 1800;refresher=uas
  179. Content-Type: application/sdp
  180. Content-Length: 290
  181. v=0
  182. s=Asterisk PBX 16.1.1
  183. t=0 0
  184. a=rtpmap:0 PCMU/8000
  185. a=rtpmap:3 GSM/8000
  186. a=fmtp:101 0-16
  187. a=sendrecv
  188. --- (14 headers 13 lines) ---
  189. Found RTP audio format 8
  190. Found RTP audio format 101
  191. Found audio description format PCMA for ID 8
  192. Found audio description format telephone-event for ID 101
  193. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw gsm alaw)/video=(nothing)/text=(nothing), combined - (ulaw alaw gsm)
  194. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  195. > 0x7f780401b140 -- Strict RTP learning after remote address set to: 104.156.255.135:16364
  196. sip_route_dump: route/path hop: <sip:100@104.156.255.135:5060>
  197. set_destination: Parsing <sip:100@104.156.255.135:5060> for address/port to send to
  198. set_destination: set destination to 104.156.255.135:5060
  199. ACK sip:100@104.156.255.135:5060 SIP/2.0
  200. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK1856f145
  201. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  202. Contact: <sip:14103101234@45.76.62.92:5060>
  203. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  204. User-Agent: Asterisk PBX 16.1.1
  205. ---
  206. -- SIP/server-b-0000003d answered SIP/71.200.12.159-0000003c
  207. -- Channel SIP/server-b-0000003d joined 'simple_bridge' basic-bridge <5aa11cc5-f01c-4a3a-a5bb-0a5a508fe801>
  208. -- Channel SIP/71.200.12.159-0000003c joined 'simple_bridge' basic-bridge <5aa11cc5-f01c-4a3a-a5bb-0a5a508fe801>
  209. > 0x7f780401b140 -- Strict RTP switching to RTP target address 104.156.255.135:16364 as source
  210. Reliably Transmitting (no NAT) to 142.54.168.146:5060:
  211. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK5c26716d
  212. From: 'asterisk' <sip:asterisk@45.76.62.92>;tag=as4476ac48
  213. Contact: <sip:asterisk@45.76.62.92:5060>
  214. Call-ID: 175a85fb43e5a0b36149ddb82f9189a6@45.76.62.92:5060
  215. User-Agent: Asterisk PBX 16.1.1
  216. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  217. Content-Length: 0
  218. SIP/2.0 200 OK
  219. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK5c26716d;received=45.76.62.92
  220. From: 'asterisk' <sip:asterisk@45.76.62.92>;tag=as4476ac48
  221. Call-ID: 175a85fb43e5a0b36149ddb82f9189a6@45.76.62.92:5060
  222. Server: Asterisk PBX 16.1.1
  223. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  224. Contact: <sip:142.54.168.146:5060>
  225. Content-Length: 0
  226. <------------->
  227. Really destroying SIP dialog '175a85fb43e5a0b36149ddb82f9189a6@45.76.62.92:5060' Method: OPTIONS
  228. SIP/2.0 404 Not Found
  229. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  230. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  231. CSeq: 1 INVITE
  232. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  233. Content-Length: 0
  234. INVITE sip:200@45.76.62.92:5060 SIP/2.0
  235. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674
  236. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  237. Contact: <sip:14103101234@104.156.255.135:5060>
  238. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  239. User-Agent: Asterisk PBX 16.1.1
  240. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  241. Content-Type: application/sdp
  242. o=root 150099083 150099083 IN IP4 104.156.255.135
  243. c=IN IP4 104.156.255.135
  244. m=audio 14390 RTP/AVP 0 8 3 101
  245. a=rtpmap:8 PCMA/8000
  246. a=rtpmap:101 telephone-event/8000
  247. a=maxptime:150
  248. <------------->
  249. Sending to 104.156.255.135:5060 (no NAT)
  250. Using INVITE request as basis request - 37b98421766328a1309136135860925f@104.156.255.135:5060
  251. Found peer 'server-b' for '14103101234' from 104.156.255.135:5060
  252. Found RTP audio format 0
  253. Found RTP audio format 3
  254. Found audio description format PCMU for ID 0
  255. Found audio description format GSM for ID 3
  256. Found audio description format telephone-event for ID 101
  257. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw gsm alaw)/video=(nothing)/text=(nothing), combined - (ulaw alaw gsm)
  258. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  259. > 0x7f77f40119c0 -- Strict RTP learning after remote address set to: 104.156.255.135:14390
  260. Looking for 200 in server-b (domain 45.76.62.92)
  261. sip_route_dump: route/path hop: <sip:14103101234@104.156.255.135:5060>
  262. <--- Transmitting (no NAT) to 104.156.255.135:5060 --->
  263. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674;received=104.156.255.135
  264. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  265. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  266. Server: Asterisk PBX 16.1.1
  267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  268. Session-Expires: 1800;refresher=uas
  269. Content-Length: 0
  270. -- Executing [200@server-b:1] Playback('SIP/server-b-0000003e', 'beep') in new stack
  271. Adding codec ulaw to SDP
  272. Adding codec gsm to SDP
  273. <--- Reliably Transmitting (no NAT) to 104.156.255.135:5060 --->
  274. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674;received=104.156.255.135
  275. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  276. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  277. Server: Asterisk PBX 16.1.1
  278. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  279. Session-Expires: 1800;refresher=uas
  280. Content-Type: application/sdp
  281. Content-Length: 284
  282. v=0
  283. s=Asterisk PBX 16.1.1
  284. t=0 0
  285. a=rtpmap:0 PCMU/8000
  286. a=rtpmap:3 GSM/8000
  287. a=fmtp:101 0-16
  288. a=sendrecv
  289. <------------>
  290. <--- SIP read from UDP:104.156.255.135:5060 --->
  291. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK3a3760d5
  292. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  293. Contact: <sip:14103101234@104.156.255.135:5060>
  294. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  295. User-Agent: Asterisk PBX 16.1.1
  296. --- (10 headers 0 lines) ---
  297. > 0x7f77f40119c0 -- Strict RTP switching to RTP target address 104.156.255.135:14390 as source
  298. -- <SIP/server-b-0000003e> Playing 'beep.gsm' (language 'en')
  299. -- Executing [200@server-b:2] Playback('SIP/server-b-0000003e', 'beep') in new stack
  300. -- <SIP/server-b-0000003e> Playing 'beep.gsm' (language 'en')
  301. -- Executing [200@server-b:3] Playback('SIP/server-b-0000003e', 'beep') in new stack
  302. -- <SIP/server-b-0000003e> Playing 'beep.gsm' (language 'en')
  303. > 0x7f780401b140 -- Strict RTP learning complete - Locking on source address 104.156.255.135:16364
  304. -- Executing [200@server-b:4] Playback('SIP/server-b-0000003e', 'beep') in new stack
  305. -- <SIP/server-b-0000003e> Playing 'beep.gsm' (language 'en')
  306. -- Executing [200@server-b:5] Playback('SIP/server-b-0000003e', 'beep') in new stack
  307. -- <SIP/server-b-0000003e> Playing 'beep.gsm' (language 'en')
  308. -- Executing [200@server-b:6] Wait('SIP/server-b-0000003e', '10') in new stack
  309. SIP/2.0 404 Not Found
  310. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  311. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  312. CSeq: 1 INVITE
  313. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  314. Content-Length: 0
  315. > 0x7f77f40119c0 -- Strict RTP learning complete - Locking on source address 104.156.255.135:14390
  316. SIP/2.0 404 Not Found
  317. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  318. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  319. CSeq: 1 INVITE
  320. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  321. Content-Length: 0
  322. Retransmitting #8 (no NAT) to 185.53.91.27:5070:
  323. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  324. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  325. CSeq: 1 INVITE
  326. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  327. Content-Length: 0
  328. -- Executing [200@server-b:7] Hangup('SIP/server-b-0000003e', ') in new stack
  329. Spawn extension (server-b, 200, 7) exited non-zero on 'SIP/server-b-0000003e'
  330. Scheduling destruction of SIP dialog '37b98421766328a1309136135860925f@104.156.255.135:5060' in 32000 ms (Method: ACK)
  331. set_destination: Parsing <sip:14103101234@104.156.255.135:5060> for address/port to send to
  332. set_destination: set destination to 104.156.255.135:5060
  333. Reliably Transmitting (no NAT) to 104.156.255.135:5060:
  334. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK63554106
  335. From: <sip:200@45.76.62.92:5060>;tag=as07a68422
  336. To: <sip:14103101234@104.156.255.135>;tag=as698c269f
  337. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  338. User-Agent: Asterisk PBX 16.1.1
  339. X-Asterisk-HangupCauseCode: 16
  340. ---
  341. <--- SIP read from UDP:104.156.255.135:5060 --->
  342. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK63554106;received=45.76.62.92
  343. To: <sip:14103101234@104.156.255.135>;tag=as698c269f
  344. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  345. Server: Asterisk PBX 16.1.1
  346. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  347. Content-Length: 0
  348. <------------->
  349. SIP Response message for INCOMING dialog BYE arrived
  350. Really destroying SIP dialog '37b98421766328a1309136135860925f@104.156.255.135:5060' Method: ACK
  351. <--- SIP read from UDP:104.156.255.135:5060 --->
  352. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK1f7743b7
  353. From: <sip:100@104.156.255.135:5060>;tag=as324a6628
  354. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  355. User-Agent: Asterisk PBX 16.1.1
  356. X-Asterisk-HangupCauseCode: 16
  357. --- (11 headers 0 lines) ---
  358. Scheduling destruction of SIP dialog '04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060' in 32000 ms (Method: BYE)
  359. <--- Transmitting (no NAT) to 104.156.255.135:5060 --->
  360. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK1f7743b7;received=104.156.255.135
  361. From: <sip:100@104.156.255.135:5060>;tag=as324a6628
  362. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  363. Server: Asterisk PBX 16.1.1
  364. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  365. Content-Length: 0
  366. -- Channel SIP/server-b-0000003d left 'simple_bridge' basic-bridge <5aa11cc5-f01c-4a3a-a5bb-0a5a508fe801>
  367. -- Channel SIP/71.200.12.159-0000003c left 'simple_bridge' basic-bridge <5aa11cc5-f01c-4a3a-a5bb-0a5a508fe801>
  368. Spawn extension (public, 100, 3) exited non-zero on 'SIP/71.200.12.159-0000003c'
  369. Scheduling destruction of SIP dialog 'chcjTcZm04RsObIGC7dLVg..' in 32000 ms (Method: ACK)
  370. set_destination: Parsing <sip:14103101234@71.200.12.159:61092;transport=UDP> for address/port to send to
  371. set_destination: set destination to 71.200.12.159:61092
  372. Reliably Transmitting (no NAT) to 71.200.12.159:61092:
  373. BYE sip:14103101234@71.200.12.159:61092;transport=UDP SIP/2.0
  374. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK5a6abf3c;rport
  375. From: <sip:100@45.76.62.92:5060;transport=UDP>;tag=as79a46b5c
  376. To: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  377. CSeq: 102 BYE
  378. X-Asterisk-HangupCause: Normal Clearing
  379. Content-Length: 0
  380. SIP/2.0 200 OK
  381. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK5a6abf3c;rport=5060
  382. Contact: <sip:14103101234@71.200.12.159:61092;transport=UDP>
  383. To: <sip:14103101234@71.200.12.159;transport=UDP>;tag=f76bb00f
  384. From: <sip:100@45.76.62.92:5060;transport=UDP>;tag=as79a46b5c
  385. CSeq: 102 BYE
  386. Content-Length: 0
  387. <------------->
  388. SIP Response message for INCOMING dialog BYE arrived
  389. Really destroying SIP dialog 'chcjTcZm04RsObIGC7dLVg..' Method: ACK
  390. SIP/2.0 404 Not Found
  391. Via: SIP/2.0/UDP 185.53.91.27:5070;branch=z9hG4bK-1302bfada3c0539b1c4bd36b57286fc8;received=185.53.91.27;rport=5070
  392. To: 22011972592315281<sip:22011972592315281@45.76.62.92>;tag=as202a317f
  393. CSeq: 1 INVITE
  394. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  395. Content-Length: 0
  396. ############
  397. ############
  398. <--- SIP read from UDP:45.76.62.92:5060 --->
  399. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a
  400. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  401. Contact: <sip:14103101234@45.76.62.92:5060>
  402. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  403. User-Agent: Asterisk PBX 16.1.1
  404. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  405. Content-Type: application/sdp
  406. o=root 407118028 407118028 IN IP4 45.76.62.92
  407. c=IN IP4 45.76.62.92
  408. m=audio 17804 RTP/AVP 0 8 3 101
  409. a=rtpmap:8 PCMA/8000
  410. a=rtpmap:101 telephone-event/8000
  411. a=maxptime:150
  412. <------------->
  413. Sending to 45.76.62.92:5060 (no NAT)
  414. Using INVITE request as basis request - 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  415. Found peer 'server-a' for '14103101234' from 45.76.62.92:5060
  416. Found RTP audio format 0
  417. Found RTP audio format 3
  418. Found audio description format PCMU for ID 0
  419. Found audio description format GSM for ID 3
  420. Found audio description format telephone-event for ID 101
  421. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw gsm alaw)/video=(nothing)/text=(nothing), combined - (ulaw alaw gsm)
  422. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  423. > 0x7f715c010fd0 -- Strict RTP learning after remote address set to: 45.76.62.92:17804
  424. Looking for 100 in server-a (domain 104.156.255.135)
  425. sip_route_dump: route/path hop: <sip:14103101234@45.76.62.92:5060>
  426. <--- Transmitting (no NAT) to 45.76.62.92:5060 --->
  427. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a;received=45.76.62.92
  428. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  429. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  430. Server: Asterisk PBX 16.1.1
  431. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  432. Session-Expires: 1800;refresher=uas
  433. Content-Length: 0
  434. -- Executing [100@server-a:1] Playback('SIP/server-a-00000026', 'agent-newlocation') in new stack
  435. Adding codec ulaw to SDP
  436. Adding codec gsm to SDP
  437. <--- Reliably Transmitting (no NAT) to 45.76.62.92:5060 --->
  438. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK7bb51d8a;received=45.76.62.92
  439. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  440. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  441. Server: Asterisk PBX 16.1.1
  442. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  443. Session-Expires: 1800;refresher=uas
  444. Content-Type: application/sdp
  445. Content-Length: 290
  446. v=0
  447. s=Asterisk PBX 16.1.1
  448. t=0 0
  449. a=rtpmap:0 PCMU/8000
  450. a=rtpmap:3 GSM/8000
  451. a=fmtp:101 0-16
  452. a=sendrecv
  453. <------------>
  454. <--- SIP read from UDP:45.76.62.92:5060 --->
  455. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK1856f145
  456. From: <sip:14103101234@45.76.62.92>;tag=as3c7d3421
  457. Contact: <sip:14103101234@45.76.62.92:5060>
  458. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  459. User-Agent: Asterisk PBX 16.1.1
  460. --- (10 headers 0 lines) ---
  461. > 0x7f715c010fd0 -- Strict RTP switching to RTP target address 45.76.62.92:17804 as source
  462. -- <SIP/server-a-00000026> Playing 'agent-newlocation.gsm' (language 'en')
  463. -- Executing [100@server-a:2] Playback('SIP/server-a-00000026', 'beep') in new stack
  464. -- <SIP/server-a-00000026> Playing 'beep.gsm' (language 'en')
  465. -- Executing [100@server-a:3] Dial('SIP/server-a-00000026', 'SIP/200@server-a,Th') in new stack
  466. Audio is at 14390
  467. Adding codec alaw to SDP
  468. Adding non-codec 0x1 (telephone-event) to SDP
  469. Reliably Transmitting (no NAT) to 45.76.62.92:5060:
  470. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674
  471. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  472. Contact: <sip:14103101234@104.156.255.135:5060>
  473. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  474. User-Agent: Asterisk PBX 16.1.1
  475. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  476. Content-Type: application/sdp
  477. o=root 150099083 150099083 IN IP4 104.156.255.135
  478. c=IN IP4 104.156.255.135
  479. m=audio 14390 RTP/AVP 0 8 3 101
  480. a=rtpmap:8 PCMA/8000
  481. a=rtpmap:101 telephone-event/8000
  482. a=maxptime:150
  483. -- Called SIP/200@server-a
  484. <--- SIP read from UDP:45.76.62.92:5060 --->
  485. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674;received=104.156.255.135
  486. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  487. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  488. Server: Asterisk PBX 16.1.1
  489. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  490. Session-Expires: 1800;refresher=uas
  491. Content-Length: 0
  492. <------------->
  493. SIP/2.0 200 OK
  494. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK103c7674;received=104.156.255.135
  495. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  496. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  497. Server: Asterisk PBX 16.1.1
  498. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  499. Session-Expires: 1800;refresher=uas
  500. Content-Type: application/sdp
  501. Content-Length: 284
  502. v=0
  503. s=Asterisk PBX 16.1.1
  504. t=0 0
  505. a=rtpmap:0 PCMU/8000
  506. a=rtpmap:3 GSM/8000
  507. a=fmtp:101 0-16
  508. a=sendrecv
  509. --- (14 headers 13 lines) ---
  510. Found RTP audio format 8
  511. Found RTP audio format 101
  512. Found audio description format PCMA for ID 8
  513. Found audio description format telephone-event for ID 101
  514. Capabilities: us - (ulaw alaw gsm h263), peer - audio=(ulaw gsm alaw)/video=(nothing)/text=(nothing), combined - (ulaw alaw gsm)
  515. Non-codec capabilities (dtmf): us - 0x1 (telephone-event ), peer - 0x1 (telephone-event ), combined - 0x1 (telephone-event )
  516. > 0x7f7154010270 -- Strict RTP learning after remote address set to: 45.76.62.92:12740
  517. sip_route_dump: route/path hop: <sip:200@45.76.62.92:5060>
  518. set_destination: Parsing <sip:200@45.76.62.92:5060> for address/port to send to
  519. set_destination: set destination to 45.76.62.92:5060
  520. ACK sip:200@45.76.62.92:5060 SIP/2.0
  521. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK3a3760d5
  522. From: <sip:14103101234@104.156.255.135>;tag=as698c269f
  523. Contact: <sip:14103101234@104.156.255.135:5060>
  524. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  525. User-Agent: Asterisk PBX 16.1.1
  526. ---
  527. -- SIP/server-a-00000027 answered SIP/server-a-00000026
  528. -- Channel SIP/server-a-00000027 joined 'simple_bridge' basic-bridge <da06b311-6c72-43bc-a102-6dffbc1c9cdc>
  529. -- Channel SIP/server-a-00000026 joined 'simple_bridge' basic-bridge <da06b311-6c72-43bc-a102-6dffbc1c9cdc>
  530. > 0x7f7154010270 -- Strict RTP switching to RTP target address 45.76.62.92:12740 as source
  531. > 0x7f715c010fd0 -- Strict RTP learning complete - Locking on source address 45.76.62.92:17804
  532. Reliably Transmitting (no NAT) to 142.54.168.146:5060:
  533. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK15d7f85c
  534. From: 'asterisk' <sip:asterisk@104.156.255.135>;tag=as48034b22
  535. Contact: <sip:asterisk@104.156.255.135:5060>
  536. Call-ID: 3eaf63ba573e276469dc8f37171d6ed9@104.156.255.135:5060
  537. User-Agent: Asterisk PBX 16.1.1
  538. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  539. Content-Length: 0
  540. SIP/2.0 200 OK
  541. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK15d7f85c;received=104.156.255.135
  542. From: 'asterisk' <sip:asterisk@104.156.255.135>;tag=as48034b22
  543. Call-ID: 3eaf63ba573e276469dc8f37171d6ed9@104.156.255.135:5060
  544. Server: Asterisk PBX 16.1.1
  545. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  546. Contact: <sip:142.54.168.146:5060>
  547. Content-Length: 0
  548. <------------->
  549. Really destroying SIP dialog '3eaf63ba573e276469dc8f37171d6ed9@104.156.255.135:5060' Method: OPTIONS
  550. <--- SIP read from UDP:45.76.62.92:5060 --->
  551. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK63554106
  552. From: <sip:200@45.76.62.92:5060>;tag=as07a68422
  553. To: <sip:14103101234@104.156.255.135>;tag=as698c269f
  554. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  555. User-Agent: Asterisk PBX 16.1.1
  556. X-Asterisk-HangupCauseCode: 16
  557. --- (11 headers 0 lines) ---
  558. Scheduling destruction of SIP dialog '37b98421766328a1309136135860925f@104.156.255.135:5060' in 32000 ms (Method: BYE)
  559. <--- Transmitting (no NAT) to 45.76.62.92:5060 --->
  560. Via: SIP/2.0/UDP 45.76.62.92:5060;branch=z9hG4bK63554106;received=45.76.62.92
  561. To: <sip:14103101234@104.156.255.135>;tag=as698c269f
  562. Call-ID: 37b98421766328a1309136135860925f@104.156.255.135:5060
  563. Server: Asterisk PBX 16.1.1
  564. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  565. Content-Length: 0
  566. -- Channel SIP/server-a-00000027 left 'simple_bridge' basic-bridge <da06b311-6c72-43bc-a102-6dffbc1c9cdc>
  567. -- Channel SIP/server-a-00000026 left 'simple_bridge' basic-bridge <da06b311-6c72-43bc-a102-6dffbc1c9cdc>
  568. Spawn extension (server-a, 100, 3) exited non-zero on 'SIP/server-a-00000026'
  569. Scheduling destruction of SIP dialog '04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060' in 32000 ms (Method: ACK)
  570. set_destination: Parsing <sip:14103101234@45.76.62.92:5060> for address/port to send to
  571. set_destination: set destination to 45.76.62.92:5060
  572. Reliably Transmitting (no NAT) to 45.76.62.92:5060:
  573. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK1f7743b7
  574. From: <sip:100@104.156.255.135:5060>;tag=as324a6628
  575. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  576. User-Agent: Asterisk PBX 16.1.1
  577. X-Asterisk-HangupCauseCode: 16
  578. ---
  579. <--- SIP read from UDP:45.76.62.92:5060 --->
  580. Via: SIP/2.0/UDP 104.156.255.135:5060;branch=z9hG4bK1f7743b7;received=104.156.255.135
  581. From: <sip:100@104.156.255.135:5060>;tag=as324a6628
  582. Call-ID: 04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060
  583. Server: Asterisk PBX 16.1.1
  584. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  585. Content-Length: 0
  586. <------------->
  587. SIP Response message for INCOMING dialog BYE arrived
  588. Really destroying SIP dialog '04383e9760dfbba265c8f7655d32ee7a@45.76.62.92:5060' Method: ACK

This is how we protect your privacy. We’re committed to keeping your personal information safe. That’s why we innovate ways to safeguard your privacy on your device, why we’re up front about how we personalize your experience, and why we equip developers with the best tools to protect your data. Rtp-timeout= RTP source timeout (sec) How long to wait for any packet before a source is expired.rtp-max-dropout= Maximum RTP sequence number dropout RTP packets will be discarded if they are too much ahead (i.e. In the future) by this many packets from the last received packet. Wireless Application Protocol (WAP) is a technical standard for accessing information over a mobile wireless network.A WAP browser is a web browser for mobile devices such as mobile phones that uses the protocol. Introduced with much hype in 1999, WAP achieved some popularity in the early 2000s, but by the 2010s it had been largely superseded by more modern standards.