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Jul 23rd, 2019
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  3. centos-2gb-nbg1-1*CLI>
  4. centos-2gb-nbg1-1*CLI>
  5. centos-2gb-nbg1-1*CLI>
  6. centos-2gb-nbg1-1*CLI>
  7. centos-2gb-nbg1-1*CLI>
  8. centos-2gb-nbg1-1*CLI>
  9. centos-2gb-nbg1-1*CLI>
  10. centos-2gb-nbg1-1*CLI>
  11. centos-2gb-nbg1-1*CLI>
  12. centos-2gb-nbg1-1*CLI>
  13. <--- Received SIP request (1606 bytes) from UDP:62.44.138.87:58356 --->
  14. INVITE sip:42153189@195.201.144.187 SIP/2.0
  15. Via: SIP/2.0/UDP 10.242.74.86:58356;branch=z9hG4bK.ZivtnSRPa;rport
  16. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  17. To: sip:42153189@195.201.144.187
  18. CSeq: 20 INVITE
  19. Call-ID: O3WISc94kK
  20. Max-Forwards: 70
  21. Supported: replaces, outbound, gruu
  22. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  23. Content-Type: application/sdp
  24. Content-Length: 733
  25. Contact: <sip:6001@62.44.138.87:58356;app-id=929724111839;pn-type=firebase;pn-timeout=0;pn-tok=daHLG99Vtac:APA91bHIngd7yhkMvu_lKiSrW69mKslwrKOVKrsx26fnIlb4_8c8oiAg-rAFgvVWd7nZZqQroMlxSDWPzZgNXYgYZ9h94gqe5K7q7BHtdr5r7KCwURw0H78bvtY-stj5qcMPOaH4KTRu;pn-silent=1;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f570e6ee-e031-0040-a794-a11a705ba69a>"
  26. User-Agent: LinphoneAndroid/4.1.1 (Trololol) LinphoneSDK/4.2-8-g96666b3 (release/4.2)
  27.  
  28. v=0
  29. o=6001 2408 3060 IN IP4 10.242.74.86
  30. s=Talk
  31. c=IN IP4 10.242.74.86
  32. t=0 0
  33. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  34. m=audio 7078 RTP/AVP 96 97 98 0 8 3 9 99 18 100 102 10 11 101 103 104 105 106
  35. a=rtpmap:96 opus/48000/2
  36. a=fmtp:96 useinbandfec=1
  37. a=rtpmap:97 speex/16000
  38. a=fmtp:97 vbr=on
  39. a=rtpmap:98 speex/8000
  40. a=fmtp:98 vbr=on
  41. a=rtpmap:99 iLBC/8000
  42. a=fmtp:99 mode=30
  43. a=fmtp:18 annexb=yes
  44. a=rtpmap:100 iSAC/16000
  45. a=rtpmap:102 speex/32000
  46. a=fmtp:102 vbr=on
  47. a=rtpmap:101 telephone-event/48000
  48. a=rtpmap:103 telephone-event/16000
  49. a=rtpmap:104 telephone-event/8000
  50. a=rtpmap:105 telephone-event/32000
  51. a=rtpmap:106 telephone-event/44100
  52. a=rtcp-fb:* trr-int 1000
  53. a=rtcp-fb:* ccm tmmbr
  54.  
  55. <--- Transmitting SIP response (463 bytes) to UDP:62.44.138.87:58356 --->
  56. SIP/2.0 401 Unauthorized
  57. Via: SIP/2.0/UDP 10.242.74.86:58356;rport=58356;received=62.44.138.87;branch=z9hG4bK.ZivtnSRPa
  58. Call-ID: O3WISc94kK
  59. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  60. To: <sip:42153189@195.201.144.187>;tag=z9hG4bK.ZivtnSRPa
  61. CSeq: 20 INVITE
  62. WWW-Authenticate: Digest realm="asterisk",nonce="1563881915/2abcaa97e4753508b3e9e8a3661a1210",opaque="77ff58242d4155cc",algorithm=md5,qop="auth"
  63. Server: Asterisk PBX 16.4.1
  64. Content-Length: 0
  65.  
  66.  
  67. <--- Received SIP request (624 bytes) from UDP:62.44.138.87:58356 --->
  68. ACK sip:42153189@195.201.144.187 SIP/2.0
  69. Via: SIP/2.0/UDP 10.242.74.86:58356;branch=z9hG4bK.ZivtnSRPa;rport
  70. Call-ID: O3WISc94kK
  71. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  72. To: <sip:42153189@195.201.144.187>;tag=z9hG4bK.ZivtnSRPa
  73. Contact: <sip:6001@62.44.138.87:58356;app-id=929724111839;pn-type=firebase;pn-timeout=0;pn-tok=daHLG99Vtac:APA91bHIngd7yhkMvu_lKiSrW69mKslwrKOVKrsx26fnIlb4_8c8oiAg-rAFgvVWd7nZZqQroMlxSDWPzZgNXYgYZ9h94gqe5K7q7BHtdr5r7KCwURw0H78bvtY-stj5qcMPOaH4KTRu;pn-silent=1;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f570e6ee-e031-0040-a794-a11a705ba69a>"
  74. Max-Forwards: 70
  75. CSeq: 20 ACK
  76.  
  77.  
  78. <--- Received SIP request (1891 bytes) from UDP:62.44.138.87:58356 --->
  79. INVITE sip:42153189@195.201.144.187 SIP/2.0
  80. Via: SIP/2.0/UDP 10.242.74.86:58356;branch=z9hG4bK.pdODlBQNs;rport
  81. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  82. To: sip:42153189@195.201.144.187
  83. CSeq: 21 INVITE
  84. Call-ID: O3WISc94kK
  85. Max-Forwards: 70
  86. Supported: replaces, outbound, gruu
  87. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
  88. Content-Type: application/sdp
  89. Content-Length: 733
  90. Contact: <sip:6001@62.44.138.87:58356;app-id=929724111839;pn-type=firebase;pn-timeout=0;pn-tok=daHLG99Vtac:APA91bHIngd7yhkMvu_lKiSrW69mKslwrKOVKrsx26fnIlb4_8c8oiAg-rAFgvVWd7nZZqQroMlxSDWPzZgNXYgYZ9h94gqe5K7q7BHtdr5r7KCwURw0H78bvtY-stj5qcMPOaH4KTRu;pn-silent=1;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f570e6ee-e031-0040-a794-a11a705ba69a>"
  91. User-Agent: LinphoneAndroid/4.1.1 (Trololol) LinphoneSDK/4.2-8-g96666b3 (release/4.2)
  92. Authorization: Digest realm="asterisk", nonce="1563881915/2abcaa97e4753508b3e9e8a3661a1210", algorithm=md5, opaque="77ff58242d4155cc", username="6001", uri="sip:42153189@195.201.144.187", response="ba2431225d2836ef87f32aed34a843fa", cnonce="DU~SJPepD9Q-jBBA", nc=00000001, qop=auth
  93.  
  94. v=0
  95. o=6001 2408 3060 IN IP4 10.242.74.86
  96. s=Talk
  97. c=IN IP4 10.242.74.86
  98. t=0 0
  99. a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
  100. m=audio 7078 RTP/AVP 96 97 98 0 8 3 9 99 18 100 102 10 11 101 103 104 105 106
  101. a=rtpmap:96 opus/48000/2
  102. a=fmtp:96 useinbandfec=1
  103. a=rtpmap:97 speex/16000
  104. a=fmtp:97 vbr=on
  105. a=rtpmap:98 speex/8000
  106. a=fmtp:98 vbr=on
  107. a=rtpmap:99 iLBC/8000
  108. a=fmtp:99 mode=30
  109. a=fmtp:18 annexb=yes
  110. a=rtpmap:100 iSAC/16000
  111. a=rtpmap:102 speex/32000
  112. a=fmtp:102 vbr=on
  113. a=rtpmap:101 telephone-event/48000
  114. a=rtpmap:103 telephone-event/16000
  115. a=rtpmap:104 telephone-event/8000
  116. a=rtpmap:105 telephone-event/32000
  117. a=rtpmap:106 telephone-event/44100
  118. a=rtcp-fb:* trr-int 1000
  119. a=rtcp-fb:* ccm tmmbr
  120.  
  121. == Setting global variable 'SIPDOMAIN' to '195.201.144.187'
  122. <--- Transmitting SIP response (289 bytes) to UDP:62.44.138.87:58356 --->
  123. SIP/2.0 100 Trying
  124. Via: SIP/2.0/UDP 10.242.74.86:58356;rport=58356;received=62.44.138.87;branch=z9hG4bK.pdODlBQNs
  125. Call-ID: O3WISc94kK
  126. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  127. To: <sip:42153189@195.201.144.187>
  128. CSeq: 21 INVITE
  129. Server: Asterisk PBX 16.4.1
  130. Content-Length: 0
  131.  
  132.  
  133. -- Executing [42153189@internal:1] Dial("PJSIP/6001-00000007", "PJSIP/004542153189@mytrunk,30,tr") in new stack
  134. <--- Transmitting SIP request (921 bytes) to UDP:94.75.247.45:5060 --->
  135. INVITE sip:004542153189@localphone.com:5060 SIP/2.0
  136. Via: SIP/2.0/UDP 195.201.144.187:5060;rport;branch=z9hG4bKPj35899d71-793d-45ca-bcaa-1ffd02ae2506
  137. From: <sip:6001@195.201.144.187>;tag=43d8782b-249a-4b5a-8fce-a6aa9bbb9527
  138. To: <sip:004542153189@localphone.com>
  139. Contact: <sip:asterisk@195.201.144.187:5060>
  140. Call-ID: 8895851d-45d2-477c-b499-b101fe92da95
  141. CSeq: 1529 INVITE
  142. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  143. Supported: 100rel, timer, replaces, norefersub
  144. Session-Expires: 1800
  145. Min-SE: 90
  146. Max-Forwards: 70
  147. User-Agent: Asterisk PBX 16.4.1
  148. Content-Type: application/sdp
  149. Content-Length: 243
  150.  
  151. v=0
  152. o=- 1165497964 1165497964 IN IP4 195.201.144.187
  153. s=Asterisk
  154. c=IN IP4 195.201.144.187
  155. t=0 0
  156. m=audio 10698 RTP/AVP 0 101
  157. a=rtpmap:0 PCMU/8000
  158. a=rtpmap:101 telephone-event/8000
  159. a=fmtp:101 0-16
  160. a=ptime:20
  161. a=maxptime:150
  162. a=sendrecv
  163.  
  164. -- Called PJSIP/004542153189@mytrunk
  165. <--- Transmitting SIP response (479 bytes) to UDP:62.44.138.87:58356 --->
  166. SIP/2.0 180 Ringing
  167. Via: SIP/2.0/UDP 10.242.74.86:58356;rport=58356;received=62.44.138.87;branch=z9hG4bK.pdODlBQNs
  168. Call-ID: O3WISc94kK
  169. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  170. To: <sip:42153189@195.201.144.187>;tag=5b90f6f7-e7b9-4162-8dae-63a44bb16cb5
  171. CSeq: 21 INVITE
  172. Server: Asterisk PBX 16.4.1
  173. Contact: <sip:195.201.144.187:5060>
  174. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  175. Content-Length: 0
  176.  
  177.  
  178. <--- Received SIP response (379 bytes) from UDP:94.75.247.45:5060 --->
  179. SIP/2.0 100 Giving a try...
  180. Via: SIP/2.0/UDP 195.201.144.187:5060;rport=5060;branch=z9hG4bKPj35899d71-793d-45ca-bcaa-1ffd02ae2506
  181. From: <sip:6001@195.201.144.187>;tag=43d8782b-249a-4b5a-8fce-a6aa9bbb9527
  182. To: <sip:004542153189@localphone.com>
  183. Call-ID: 8895851d-45d2-477c-b499-b101fe92da95
  184. CSeq: 1529 INVITE
  185. Server: OpenSER (1.2.2-notls (x86_64/linux))
  186. Content-Length: 0
  187.  
  188.  
  189. <--- Received SIP response (420 bytes) from UDP:94.75.247.45:5060 --->
  190. SIP/2.0 403 GW call denied
  191. Via: SIP/2.0/UDP 195.201.144.187:5060;rport=5060;branch=z9hG4bKPj35899d71-793d-45ca-bcaa-1ffd02ae2506
  192. From: <sip:6001@195.201.144.187>;tag=43d8782b-249a-4b5a-8fce-a6aa9bbb9527
  193. To: <sip:004542153189@localphone.com>;tag=9399de1ef8c379d4c914a855a096e8ba.1464
  194. Call-ID: 8895851d-45d2-477c-b499-b101fe92da95
  195. CSeq: 1529 INVITE
  196. Server: OpenSER (1.2.2-notls (x86_64/linux))
  197. Content-Length: 0
  198.  
  199.  
  200. <--- Transmitting SIP request (440 bytes) to UDP:94.75.247.45:5060 --->
  201. ACK sip:004542153189@localphone.com:5060 SIP/2.0
  202. Via: SIP/2.0/UDP 195.201.144.187:5060;rport;branch=z9hG4bKPj35899d71-793d-45ca-bcaa-1ffd02ae2506
  203. From: <sip:6001@195.201.144.187>;tag=43d8782b-249a-4b5a-8fce-a6aa9bbb9527
  204. To: <sip:004542153189@localphone.com>;tag=9399de1ef8c379d4c914a855a096e8ba.1464
  205. Call-ID: 8895851d-45d2-477c-b499-b101fe92da95
  206. CSeq: 1529 ACK
  207. Max-Forwards: 70
  208. User-Agent: Asterisk PBX 16.4.1
  209. Content-Length: 0
  210.  
  211.  
  212. == Everyone is busy/congested at this time (1:0/0/1)
  213. -- Auto fallthrough, channel 'PJSIP/6001-00000007' status is 'CHANUNAVAIL'
  214. <--- Transmitting SIP response (478 bytes) to UDP:62.44.138.87:58356 --->
  215. SIP/2.0 503 Service Unavailable
  216. Via: SIP/2.0/UDP 10.242.74.86:58356;rport=58356;received=62.44.138.87;branch=z9hG4bK.pdODlBQNs
  217. Call-ID: O3WISc94kK
  218. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  219. To: <sip:42153189@195.201.144.187>;tag=5b90f6f7-e7b9-4162-8dae-63a44bb16cb5
  220. CSeq: 21 INVITE
  221. Server: Asterisk PBX 16.4.1
  222. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  223. Reason: Q.850;cause=21
  224. Content-Length: 0
  225.  
  226.  
  227. <--- Received SIP request (643 bytes) from UDP:62.44.138.87:58356 --->
  228. ACK sip:42153189@195.201.144.187 SIP/2.0
  229. Via: SIP/2.0/UDP 10.242.74.86:58356;branch=z9hG4bK.pdODlBQNs;rport
  230. Call-ID: O3WISc94kK
  231. From: <sip:6001@195.201.144.187>;tag=v4Qd0ff1w
  232. To: <sip:42153189@195.201.144.187>;tag=5b90f6f7-e7b9-4162-8dae-63a44bb16cb5
  233. Contact: <sip:6001@62.44.138.87:58356;app-id=929724111839;pn-type=firebase;pn-timeout=0;pn-tok=daHLG99Vtac:APA91bHIngd7yhkMvu_lKiSrW69mKslwrKOVKrsx26fnIlb4_8c8oiAg-rAFgvVWd7nZZqQroMlxSDWPzZgNXYgYZ9h94gqe5K7q7BHtdr5r7KCwURw0H78bvtY-stj5qcMPOaH4KTRu;pn-silent=1;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f570e6ee-e031-0040-a794-a11a705ba69a>"
  234. Max-Forwards: 70
  235. CSeq: 21 ACK
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