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  1. *CLI> core set verbose 10
  2. Console verbose is still 10.
  3. *CLI> core set debug 10
  4. Core debug is still 10.
  5. *CLI> sip set debug on
  6. SIP Debugging enabled
  7. *CLI> [Jun 21 15:22:42] DEBUG[1774]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
  8. [Jun 21 15:22:42] DEBUG[1766]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 3
  9. [Jun 21 15:22:42] DEBUG[1775]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
  10. [Jun 21 15:22:42] DEBUG[1772]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
  11. [Jun 21 15:22:42] DEBUG[1773]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
  12. [Jun 21 15:22:42] DEBUG[1766]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 4
  13. [Jun 21 15:22:42] DEBUG[1766]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 1
  14. [Jun 21 15:22:42] DEBUG[1766]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 2
  15. [Jun 21 15:22:43] DEBUG[1771]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying.
  16. [Jun 21 15:22:43] DEBUG[1766]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 0
  17.  
  18. <--- SIP read from UDP:212.71.138.50:62189 --->
  19. INVITE sip:111@operation.voipex.io SIP/2.0
  20. Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj990172590a41455ba480a9fd486f2300
  21. Max-Forwards: 70
  22. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  23. To: <sip:111@operation.voipex.io>
  24. Contact: <sip:400@212.71.138.50:62189;ob>
  25. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  26. CSeq: 8446 INVITE
  27. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  28. Supported: replaces, 100rel, timer, norefersub
  29. Session-Expires: 1800
  30. Min-SE: 90
  31. User-Agent: MicroSIP/3.15.4
  32. Content-Type: application/sdp
  33. Content-Length: 365
  34.  
  35. v=0
  36. o=- 3707047363 3707047363 IN IP4 212.71.138.50
  37. s=pjmedia
  38. b=AS:84
  39. t=0 0
  40. a=X-nat:0
  41. m=audio 4012 RTP/AVP 123 8 0 101
  42. c=IN IP4 212.71.138.50
  43. b=TIAS:64000
  44. a=rtcp:4013 IN IP4 10.128.3.164
  45. a=sendrecv
  46. a=rtpmap:123 opus/48000/2
  47. a=fmtp:123 maxplaybackrate=16000
  48. a=rtpmap:8 PCMA/8000
  49. a=rtpmap:0 PCMU/8000
  50. a=rtpmap:101 telephone-event/8000
  51. a=fmtp:101 0-16
  52. <------------->
  53. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 0 [ 42]: INVITE sip:111@operation.voipex.io SIP/2.0
  54. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 1 [ 91]: Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj990172590a41455ba480a9fd486f2300
  55. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 2 [ 16]: Max-Forwards: 70
  56. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 3 [ 72]: From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  57. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 4 [ 33]: To: <sip:111@operation.voipex.io>
  58. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 5 [ 41]: Contact: <sip:400@212.71.138.50:62189;ob>
  59. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 6 [ 41]: Call-ID: 07a1addc2d684b28a5257ea364141cb9
  60. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 7 [ 17]: CSeq: 8446 INVITE
  61. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  62. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub
  63. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 10 [ 21]: Session-Expires: 1800
  64. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 11 [ 10]: Min-SE: 90
  65. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 12 [ 27]: User-Agent: MicroSIP/3.15.4
  66. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 13 [ 29]: Content-Type: application/sdp
  67. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 14 [ 19]: Content-Length: 365
  68. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 15 [ 0]:
  69. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 0 [ 3]: v=0
  70. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 1 [ 46]: o=- 3707047363 3707047363 IN IP4 212.71.138.50
  71. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 2 [ 9]: s=pjmedia
  72. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 3 [ 7]: b=AS:84
  73. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 4 [ 5]: t=0 0
  74. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 5 [ 9]: a=X-nat:0
  75. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 6 [ 32]: m=audio 4012 RTP/AVP 123 8 0 101
  76. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 7 [ 22]: c=IN IP4 212.71.138.50
  77. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 8 [ 12]: b=TIAS:64000
  78. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 9 [ 31]: a=rtcp:4013 IN IP4 10.128.3.164
  79. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 10 [ 10]: a=sendrecv
  80. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 11 [ 25]: a=rtpmap:123 opus/48000/2
  81. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 12 [ 32]: a=fmtp:123 maxplaybackrate=16000
  82. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 13 [ 20]: a=rtpmap:8 PCMA/8000
  83. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 14 [ 20]: a=rtpmap:0 PCMU/8000
  84. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000
  85. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9940 parse_request: Body 16 [ 15]: a=fmtp:101 0-16
  86. --- (15 headers 17 lines) ---
  87. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9429 __find_call: = Looking for Call ID: 07a1addc2d684b28a5257ea364141cb9 (Checking From) --From tag 1d59788fdc414f7e8b4d5f9528c3bc7e --To-tag
  88. [Jun 21 15:22:44] DEBUG[1792]: acl.c:957 ast_ouraddrfor: For destination '212.71.138.50', our source address is '213.168.165.149'.
  89. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:3911 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 213.168.165.149:5060
  90. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.71.138.50:62189' into...
  91. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.71.138.50' and port '62189'.
  92. Sending to 212.71.138.50:62189 (no NAT)
  93. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9016 __sip_alloc: Allocating new SIP dialog for 07a1addc2d684b28a5257ea364141cb9 - INVITE (No RTP)
  94. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:28770 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  95. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1711 parse_sip_options: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub"
  96. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -replaces-
  97. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: replaces
  98. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -100rel-
  99. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: 100rel
  100. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -timer-
  101. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: timer
  102. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -norefersub-
  103. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: norefersub
  104. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.71.138.50:62189' into...
  105. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.71.138.50' and port '62189'.
  106. Sending to 212.71.138.50:62189 (no NAT)
  107. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:26222 handle_request_invite: Initializing initreq for method INVITE - callid 07a1addc2d684b28a5257ea364141cb9
  108. Using INVITE request as basis request - 07a1addc2d684b28a5257ea364141cb9
  109. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  110. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  111. Found peer '400' for '400' from 212.71.138.50:62189
  112.  
  113. <--- Reliably Transmitting (no NAT) to 212.71.138.50:62189 --->
  114. SIP/2.0 401 Unauthorized
  115. Via: SIP/2.0/UDP 212.71.138.50:62189;branch=z9hG4bKPj990172590a41455ba480a9fd486f2300;received=212.71.138.50;rport=62189
  116. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  117. To: <sip:111@operation.voipex.io>;tag=as7f476cde
  118. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  119. CSeq: 8446 INVITE
  120. Server: Asterisk PBX 13.16.0
  121. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  122. Supported: replaces, timer
  123. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="358b726d"
  124. Content-Length: 0
  125.  
  126.  
  127. <------------>
  128. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4267 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #4
  129. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:3754 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.71.138.50:62189
  130. Scheduling destruction of SIP dialog '07a1addc2d684b28a5257ea364141cb9' in 32000 ms (Method: INVITE)
  131.  
  132. <--- SIP read from UDP:212.71.138.50:62189 --->
  133. ACK sip:111@operation.voipex.io SIP/2.0
  134. Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj990172590a41455ba480a9fd486f2300
  135. Max-Forwards: 70
  136. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  137. To: <sip:111@operation.voipex.io>;tag=as7f476cde
  138. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  139. CSeq: 8446 ACK
  140. Content-Length: 0
  141.  
  142. <------------->
  143. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 0 [ 39]: ACK sip:111@operation.voipex.io SIP/2.0
  144. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 1 [ 91]: Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj990172590a41455ba480a9fd486f2300
  145. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 2 [ 16]: Max-Forwards: 70
  146. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 3 [ 72]: From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  147. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 4 [ 48]: To: <sip:111@operation.voipex.io>;tag=as7f476cde
  148. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 5 [ 41]: Call-ID: 07a1addc2d684b28a5257ea364141cb9
  149. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 6 [ 14]: CSeq: 8446 ACK
  150. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 7 [ 17]: Content-Length: 0
  151. --- (8 headers 0 lines) ---
  152. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9429 __find_call: = Looking for Call ID: 07a1addc2d684b28a5257ea364141cb9 (Checking From) --From tag 1d59788fdc414f7e8b4d5f9528c3bc7e --To-tag as7f476cde
  153. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:28770 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  154. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4527 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4
  155. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4538 __sip_ack: Stopping retransmission on '07a1addc2d684b28a5257ea364141cb9' of Response 8446: Match Found
  156.  
  157. <--- SIP read from UDP:212.71.138.50:62189 --->
  158. INVITE sip:111@operation.voipex.io SIP/2.0
  159. Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj5a14cf3e56bc4df39539377858163281
  160. Max-Forwards: 70
  161. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  162. To: <sip:111@operation.voipex.io>
  163. Contact: <sip:400@212.71.138.50:62189;ob>
  164. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  165. CSeq: 8447 INVITE
  166. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  167. Supported: replaces, 100rel, timer, norefersub
  168. Session-Expires: 1800
  169. Min-SE: 90
  170. User-Agent: MicroSIP/3.15.4
  171. Authorization: Digest username="400", realm="asterisk", nonce="358b726d", uri="sip:111@operation.voipex.io", response="0461a453ba6e119c3e5b27b4d8cac0c9", algorithm=MD5
  172. Content-Type: application/sdp
  173. Content-Length: 365
  174.  
  175. v=0
  176. o=- 3707047363 3707047363 IN IP4 212.71.138.50
  177. s=pjmedia
  178. b=AS:84
  179. t=0 0
  180. a=X-nat:0
  181. m=audio 4012 RTP/AVP 123 8 0 101
  182. c=IN IP4 212.71.138.50
  183. b=TIAS:64000
  184. a=rtcp:4013 IN IP4 10.128.3.164
  185. a=sendrecv
  186. a=rtpmap:123 opus/48000/2
  187. a=fmtp:123 maxplaybackrate=16000
  188. a=rtpmap:8 PCMA/8000
  189. a=rtpmap:0 PCMU/8000
  190. a=rtpmap:101 telephone-event/8000
  191. a=fmtp:101 0-16
  192. <------------->
  193. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 0 [ 42]: INVITE sip:111@operation.voipex.io SIP/2.0
  194. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 1 [ 91]: Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj5a14cf3e56bc4df39539377858163281
  195. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 2 [ 16]: Max-Forwards: 70
  196. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 3 [ 72]: From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  197. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 4 [ 33]: To: <sip:111@operation.voipex.io>
  198. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 5 [ 41]: Contact: <sip:400@212.71.138.50:62189;ob>
  199. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 6 [ 41]: Call-ID: 07a1addc2d684b28a5257ea364141cb9
  200. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 7 [ 17]: CSeq: 8447 INVITE
  201. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  202. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub
  203. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 10 [ 21]: Session-Expires: 1800
  204. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 11 [ 10]: Min-SE: 90
  205. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 12 [ 27]: User-Agent: MicroSIP/3.15.4
  206. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 13 [167]: Authorization: Digest username="400", realm="asterisk", nonce="358b726d", uri="sip:111@operation.voipex.io", response="0461a453ba6e119c3e5b27b4d8cac0c9", algorithm=MD5
  207. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 14 [ 29]: Content-Type: application/sdp
  208. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 15 [ 19]: Content-Length: 365
  209. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 16 [ 0]:
  210. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 0 [ 3]: v=0
  211. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 1 [ 46]: o=- 3707047363 3707047363 IN IP4 212.71.138.50
  212. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 2 [ 9]: s=pjmedia
  213. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 3 [ 7]: b=AS:84
  214. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 4 [ 5]: t=0 0
  215. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 5 [ 9]: a=X-nat:0
  216. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 6 [ 32]: m=audio 4012 RTP/AVP 123 8 0 101
  217. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 7 [ 22]: c=IN IP4 212.71.138.50
  218. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 8 [ 12]: b=TIAS:64000
  219. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 9 [ 31]: a=rtcp:4013 IN IP4 10.128.3.164
  220. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 10 [ 10]: a=sendrecv
  221. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 11 [ 25]: a=rtpmap:123 opus/48000/2
  222. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 12 [ 32]: a=fmtp:123 maxplaybackrate=16000
  223. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 13 [ 20]: a=rtpmap:8 PCMA/8000
  224. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 14 [ 20]: a=rtpmap:0 PCMU/8000
  225. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000
  226. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9940 parse_request: Body 16 [ 15]: a=fmtp:101 0-16
  227. --- (16 headers 17 lines) ---
  228. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9429 __find_call: = Looking for Call ID: 07a1addc2d684b28a5257ea364141cb9 (Checking From) --From tag 1d59788fdc414f7e8b4d5f9528c3bc7e --To-tag
  229. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  230. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  231. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  232. [Jun 21 15:22:44] DEBUG[1792]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  233. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:28770 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  234. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.71.138.50:62189' into...
  235. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.71.138.50' and port '62189'.
  236. Sending to 212.71.138.50:62189 (no NAT)
  237. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:26222 handle_request_invite: Initializing initreq for method INVITE - callid 07a1addc2d684b28a5257ea364141cb9
  238. Using INVITE request as basis request - 07a1addc2d684b28a5257ea364141cb9
  239. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  240. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  241. Found peer '400' for '400' from 212.71.138.50:62189
  242. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:459 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7f0f60008960'
  243. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: res_rtp_asterisk.c:3019 ast_rtp_new: Allocated port 14436 for RTP instance '0x7f0f60008960'
  244. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:476 ast_rtp_instance_new: RTP instance '0x7f0f60008960' is setup and ready to go
  245. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  246. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  247. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: res_rtp_asterisk.c:5443 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7f0f60008960'
  248. == Using SIP RTP CoS mark 5
  249. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:5799 do_setnat: Setting NAT on RTP to Off
  250. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
  251. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP o=- 3707047363 3707047363 IN IP4 212.71.138.50... OK.
  252. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED.
  253. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP b=AS:84... UNSUPPORTED OR FAILED.
  254. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
  255. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10339 process_sdp: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED.
  256. Found RTP audio format 123
  257. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:758 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 123 format for m type on 0x7f0f5817a140
  258. Found RTP audio format 8
  259. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:763 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 8 based on m type on 0x7f0f5817a140
  260. Found RTP audio format 0
  261. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:763 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 0 based on m type on 0x7f0f5817a140
  262. Found RTP audio format 101
  263. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:763 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 101 based on m type on 0x7f0f5817a140
  264. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.71.138.50' into...
  265. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.71.138.50' and port ''.
  266. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP c=IN IP4 212.71.138.50... OK.
  267. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP b=TIAS:64000... UNSUPPORTED OR FAILED.
  268. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.128.3.164... UNSUPPORTED OR FAILED.
  269. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
  270. Found audio description format opus for ID 123
  271. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=rtpmap:123 opus/48000/2... OK.
  272. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=fmtp:123 maxplaybackrate=16000... OK.
  273. Found audio description format PCMA for ID 8
  274. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
  275. Found audio description format PCMU for ID 0
  276. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  277. Found audio description format telephone-event for ID 101
  278. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  279. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:10802 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
  280. Capabilities: us - (alaw), peer - audio=(ulaw|alaw|opus)/video=(nothing)/text=(nothing), combined - (alaw)
  281. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  282. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: acl.c:957 ast_ouraddrfor: For destination '212.71.138.50', our source address is '213.168.165.149'.
  283. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: res_rtp_asterisk.c:5506 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f0f60008960'
  284. Peer audio RTP is at port 212.71.138.50:4012
  285. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:730 ast_rtp_codecs_payloads_copy: Copying payload 0 (0x7f0f60020f18) from 0x7f0f5817a140 to 0x7f0f60008b28
  286. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:730 ast_rtp_codecs_payloads_copy: Copying payload 8 (0x7f0f6001f958) from 0x7f0f5817a140 to 0x7f0f60008b28
  287. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:730 ast_rtp_codecs_payloads_copy: Copying payload 101 (0x7f0f600217a8) from 0x7f0f5817a140 to 0x7f0f60008b28
  288. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: rtp_engine.c:730 ast_rtp_codecs_payloads_copy: Copying payload 123 (0x7f0f60020218) from 0x7f0f5817a140 to 0x7f0f60008b28
  289. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: res_rtp_asterisk.c:5342 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f0f60008960'
  290. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:11097 process_sdp: We're settling with these formats: (alaw)
  291. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:26354 handle_request_invite: Checking SIP call limits for device 400
  292. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:6768 update_call_counter: Updating call counter for incoming call
  293. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:6873 update_call_counter: Call from peer '400' is 1 out of 2
  294. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  295. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  296. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'operation.voipex.io' into...
  297. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'operation.voipex.io' and port ''.
  298. Looking for 111 in test (domain operation.voipex.io)
  299. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:25819 handle_request_invite_st: Incoming INVITE with 'timer' option supported
  300. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:25829 handle_request_invite_st: INVITE also has "Session-Expires" header.
  301. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:29917 parse_session_expires: Session-Expires: 1800
  302. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:25841 handle_request_invite_st: INVITE also has "Min-SE" header.
  303. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:29887 parse_minse: Received Min-SE: 90
  304. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:8165 sip_new: *** Our native formats are (alaw)
  305. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:8166 sip_new: *** Joint capabilities are (alaw)
  306. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:8167 sip_new: *** Our capabilities are (alaw)
  307. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:8168 sip_new: *** AST_CODEC_CHOOSE formats are alaw
  308. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:8201 sip_new: This channel will not be able to handle video.
  309. sip_route_dump: route/path hop: <sip:400@212.71.138.50:62189;ob>
  310. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:26558 handle_request_invite: SIP/400-00000000: New call is still down.... Trying...
  311.  
  312. <--- Transmitting (no NAT) to 212.71.138.50:62189 --->
  313. SIP/2.0 100 Trying
  314. Via: SIP/2.0/UDP 212.71.138.50:62189;branch=z9hG4bKPj5a14cf3e56bc4df39539377858163281;received=212.71.138.50;rport=62189
  315. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  316. To: <sip:111@operation.voipex.io>
  317. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  318. CSeq: 8447 INVITE
  319. Server: Asterisk PBX 13.16.0
  320. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  321. Supported: replaces, timer
  322. Session-Expires: 1800;refresher=uas
  323. Contact: <sip:111@213.168.165.149:5060>
  324. Content-Length: 0
  325.  
  326.  
  327. <------------>
  328. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:3754 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 212.71.138.50:62189
  329. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 400
  330. [Jun 21 15:22:44] DEBUG[1777]: chan_sip.c:30307 sip_devicestate: Checking device state for peer 400
  331. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:474 do_state_change: Changing state for SIP/400 - state 2 (In use)
  332. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 400
  333. [Jun 21 15:22:44] DEBUG[1777]: chan_sip.c:30307 sip_devicestate: Checking device state for peer 400
  334. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:474 do_state_change: Changing state for SIP/400 - state 2 (In use)
  335. [Jun 21 15:22:44] DEBUG[1766]: threadpool.c:517 grow: Increasing threadpool stasis-core's size by 1
  336. [Jun 21 15:22:44] DEBUG[1796]: app_queue.c:2482 device_state_cb: Device 'SIP/400' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  337. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: pbx.c:2875 pbx_extension_helper: Launching 'NoOp'
  338. -- Executing [111@test:1] NoOp("SIP/400-00000000", "test") in new stack
  339. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: pbx.c:2875 pbx_extension_helper: Launching 'Answer'
  340. -- Executing [111@test:2] Answer("SIP/400-00000000", "") in new stack
  341. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:7413 sip_answer: SIP answering channel: SIP/400-00000000
  342. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: res_rtp_asterisk.c:3419 ast_rtp_update_source: Setting the marker bit due to a source update
  343. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:13509 add_sdp: ** Our capability: (alaw) Video flag: True Text flag: True
  344. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:13510 add_sdp: ** Our prefcodec: (nothing)
  345. Audio is at 14436
  346. Adding codec alaw to SDP
  347. Adding non-codec 0x1 (telephone-event) to SDP
  348. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:13680 add_sdp: -- Done with adding codecs to SDP
  349. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:13705 add_sdp: Setting framing on incoming call: 0
  350. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:13899 add_sdp: Done building SDP. Settling with this capability: (alaw)
  351.  
  352. <--- Reliably Transmitting (no NAT) to 212.71.138.50:62189 --->
  353. SIP/2.0 200 OK
  354. Via: SIP/2.0/UDP 212.71.138.50:62189;branch=z9hG4bKPj5a14cf3e56bc4df39539377858163281;received=212.71.138.50;rport=62189
  355. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  356. To: <sip:111@operation.voipex.io>;tag=as0152588d
  357. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  358. CSeq: 8447 INVITE
  359. Server: Asterisk PBX 13.16.0
  360. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  361. Supported: replaces, timer
  362. Session-Expires: 1800;refresher=uas
  363. Contact: <sip:111@213.168.165.149:5060>
  364. Content-Type: application/sdp
  365. Require: timer
  366. Content-Length: 246
  367.  
  368. v=0
  369. o=root 1877102108 1877102108 IN IP4 213.168.165.149
  370. s=Asterisk PBX 13.16.0
  371. c=IN IP4 213.168.165.149
  372. t=0 0
  373. m=audio 14436 RTP/AVP 8 101
  374. a=rtpmap:8 PCMA/8000
  375. a=rtpmap:101 telephone-event/8000
  376. a=fmtp:101 0-16
  377. a=maxptime:150
  378. a=sendrecv
  379.  
  380. <------------>
  381. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:4267 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #14
  382. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:3754 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.71.138.50:62189
  383. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 400
  384. [Jun 21 15:22:44] DEBUG[1777]: chan_sip.c:30307 sip_devicestate: Checking device state for peer 400
  385. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:474 do_state_change: Changing state for SIP/400 - state 2 (In use)
  386. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:29844 __start_session_timer: Session timer started: 16 - 07a1addc2d684b28a5257ea364141cb9 900000ms
  387. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: res_rtp_asterisk.c:4551 ast_rtcp_interpret: Got RTCP report of 64 bytes
  388. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: res_rtp_asterisk.c:4551 ast_rtcp_interpret: Got RTCP report of 64 bytes
  389.  
  390. <--- SIP read from UDP:212.71.138.50:62189 --->
  391. ACK sip:111@213.168.165.149:5060 SIP/2.0
  392. Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj162141a412ff43f0b66f3b3eeb6918ab
  393. Max-Forwards: 70
  394. From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  395. To: <sip:111@operation.voipex.io>;tag=as0152588d
  396. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  397. CSeq: 8447 ACK
  398. Content-Length: 0
  399.  
  400. <------------->
  401. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 0 [ 40]: ACK sip:111@213.168.165.149:5060 SIP/2.0
  402. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 1 [ 91]: Via: SIP/2.0/UDP 212.71.138.50:62189;rport;branch=z9hG4bKPj162141a412ff43f0b66f3b3eeb6918ab
  403. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 2 [ 16]: Max-Forwards: 70
  404. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 3 [ 72]: From: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  405. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 4 [ 48]: To: <sip:111@operation.voipex.io>;tag=as0152588d
  406. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 5 [ 41]: Call-ID: 07a1addc2d684b28a5257ea364141cb9
  407. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 6 [ 14]: CSeq: 8447 ACK
  408. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 7 [ 17]: Content-Length: 0
  409. --- (8 headers 0 lines) ---
  410. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9429 __find_call: = Looking for Call ID: 07a1addc2d684b28a5257ea364141cb9 (Checking From) --From tag 1d59788fdc414f7e8b4d5f9528c3bc7e --To-tag as0152588d
  411. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:28770 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  412. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4527 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14
  413. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4538 __sip_ack: Stopping retransmission on '07a1addc2d684b28a5257ea364141cb9' of Response 8447: Match Found
  414. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: res_rtp_asterisk.c:4987 ast_rtp_read: 0x7f0f600094c0 -- Probation learning mode pass with source address 212.71.138.50:4012
  415. > 0x7f0f600094c0 -- Probation passed - setting RTP source address to 212.71.138.50:4012
  416. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: pbx.c:2875 pbx_extension_helper: Launching 'Queue'
  417. -- Executing [111@test:3] Queue("SIP/400-00000000", "customer_advocate,,,,1200") in new stack
  418. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:7946 queue_exec: queue: customer_advocate, options: , url: , announce: , timeout: 1200, agi: , macro: , gosub: , rule: , position:
  419. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:7976 queue_exec: NO QUEUE_PRIO variable found. Using default.
  420. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:8028 queue_exec: queue: customer_advocate, expires: 1498052564, priority: 0
  421. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:3617 update_realtime_members: Queue customer_advocate has no realtime members defined. No need for update
  422. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:3732 join_queue: Queue 'customer_advocate' Join, Channel 'SIP/400-00000000', Position '1'
  423. [Jun 21 15:22:44] WARNING[1799][C-00000000]: config.c:3050 find_engine: Realtime mapping for 'queue_log' found to engine 'sqlite3', but the engine is not available
  424. -- Started music on hold, class 'default', on channel 'SIP/400-00000000'
  425. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: channel.c:3478 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  426. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:5299 is_our_turn: There are 0 available members.
  427. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:5317 is_our_turn: It's not our turn (SIP/400-00000000).
  428. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:3617 update_realtime_members: Queue customer_advocate has no realtime members defined. No need for update
  429. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:2213 get_member_status: PJSIP/1102 is unavailable because his device state is 'invalid'
  430. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:2213 get_member_status: PJSIP/1101 is unavailable because his device state is 'invalid'
  431. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:2213 get_member_status: PJSIP/1115 is unavailable because his device state is 'invalid'
  432. [Jun 21 15:22:44] WARNING[1799][C-00000000]: config.c:3050 find_engine: Realtime mapping for 'queue_log' found to engine 'sqlite3', but the engine is not available
  433. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: app_queue.c:4012 leave_queue: Queue 'customer_advocate' Leave, Channel 'SIP/400-00000000'
  434. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: pbx.c:4345 __ast_pbx_run: Spawn extension (test,111,3) exited non-zero on 'SIP/400-00000000'
  435. == Spawn extension (test, 111, 3) exited non-zero on 'SIP/400-00000000'
  436. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: channel.c:2582 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'SIP/400-00000000'
  437. -- Stopped music on hold on SIP/400-00000000
  438. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: channel.c:2731 ast_hangup: Hanging up channel 'SIP/400-00000000'
  439. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:7155 sip_hangup: Hangup call SIP/400-00000000, SIP callid 07a1addc2d684b28a5257ea364141cb9
  440. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:7160 sip_hangup: update_call_counter(400) - decrement call limit counter on hangup
  441. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:6768 update_call_counter: Updating call counter for incoming call
  442. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:6839 update_call_counter: Call from peer '400' removed from call limit 2
  443. Scheduling destruction of SIP dialog '07a1addc2d684b28a5257ea364141cb9' in 32000 ms (Method: ACK)
  444. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:12256 reqprep: Strict routing enforced for session 07a1addc2d684b28a5257ea364141cb9
  445. set_destination: Parsing <sip:400@212.71.138.50:62189;ob> for address/port to send to
  446. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.71.138.50:62189' into...
  447. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.71.138.50' and port '62189'.
  448. set_destination: set destination to 212.71.138.50:62189
  449. Reliably Transmitting (no NAT) to 212.71.138.50:62189:
  450. BYE sip:400@212.71.138.50:62189;ob SIP/2.0
  451. Via: SIP/2.0/UDP 213.168.165.149:5060;branch=z9hG4bK74161326;rport
  452. Max-Forwards: 70
  453. From: <sip:111@operation.voipex.io>;tag=as0152588d
  454. To: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  455. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  456. CSeq: 102 BYE
  457. User-Agent: Asterisk PBX 13.16.0
  458. Proxy-Authorization: Digest username="400", realm="asterisk", algorithm=MD5, uri="sip:operation.voipex.io", nonce="358b726d", response="34921193aba526878d39e66d28d38a56"
  459. X-Asterisk-HangupCause: Unknown
  460. X-Asterisk-HangupCauseCode: 0
  461. Content-Length: 0
  462.  
  463.  
  464. ---
  465. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:4267 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #8
  466. [Jun 21 15:22:44] DEBUG[1799][C-00000000]: chan_sip.c:3754 __sip_xmit: Trying to put 'BYE sip:400' onto UDP socket destined for 212.71.138.50:62189
  467. [Jun 21 15:22:44] DEBUG[1781]: cdr.c:1293 cdr_object_finalize: Finalized CDR for SIP/400-00000000 - start 1498051364.740059 answer 1498051364.743164 end 1498051364.762664 dispo ANSWERED
  468. [Jun 21 15:22:44] DEBUG[1796]: app_queue.c:2482 device_state_cb: Device 'Queue:customer_advocate' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
  469. [Jun 21 15:22:44] DEBUG[1796]: app_queue.c:2482 device_state_cb: Device 'Queue:customer_advocate' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  470. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 400
  471. [Jun 21 15:22:44] DEBUG[1777]: chan_sip.c:30307 sip_devicestate: Checking device state for peer 400
  472. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:474 do_state_change: Changing state for SIP/400 - state 1 (Not in use)
  473. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 400
  474. [Jun 21 15:22:44] DEBUG[1777]: chan_sip.c:30307 sip_devicestate: Checking device state for peer 400
  475. [Jun 21 15:22:44] DEBUG[1777]: devicestate.c:474 do_state_change: Changing state for SIP/400 - state 1 (Not in use)
  476. [Jun 21 15:22:44] DEBUG[1796]: app_queue.c:2482 device_state_cb: Device 'SIP/400' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  477. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:29783 do_stop_session_timer: Session timer stopped: 16 - 07a1addc2d684b28a5257ea364141cb9
  478.  
  479. <--- SIP read from UDP:212.71.138.50:62189 --->
  480. SIP/2.0 200 OK
  481. Via: SIP/2.0/UDP 213.168.165.149:5060;rport=5060;received=213.168.165.149;branch=z9hG4bK74161326
  482. Call-ID: 07a1addc2d684b28a5257ea364141cb9
  483. From: <sip:111@operation.voipex.io>;tag=as0152588d
  484. To: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  485. CSeq: 102 BYE
  486. Content-Length: 0
  487.  
  488. <------------->
  489. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 0 [ 14]: SIP/2.0 200 OK
  490. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 1 [ 96]: Via: SIP/2.0/UDP 213.168.165.149:5060;rport=5060;received=213.168.165.149;branch=z9hG4bK74161326
  491. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 2 [ 41]: Call-ID: 07a1addc2d684b28a5257ea364141cb9
  492. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 3 [ 50]: From: <sip:111@operation.voipex.io>;tag=as0152588d
  493. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 4 [ 70]: To: <sip:400@operation.voipex.io>;tag=1d59788fdc414f7e8b4d5f9528c3bc7e
  494. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 5 [ 13]: CSeq: 102 BYE
  495. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9903 parse_request: Header 6 [ 17]: Content-Length: 0
  496. --- (7 headers 0 lines) ---
  497. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:9429 __find_call: = Looking for Call ID: 07a1addc2d684b28a5257ea364141cb9 (Checking To) --From tag as0152588d --To-tag 1d59788fdc414f7e8b4d5f9528c3bc7e
  498. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4527 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8
  499. [Jun 21 15:22:44] DEBUG[1792][C-00000000]: chan_sip.c:4538 __sip_ack: Stopping retransmission on '07a1addc2d684b28a5257ea364141cb9' of Request 102: Match Found
  500. SIP Response message for INCOMING dialog BYE arrived
  501. [Jun 21 15:22:44] DEBUG[1792]: chan_sip.c:6590 sip_pvt_dtor: Destroying SIP dialog 07a1addc2d684b28a5257ea364141cb9
  502. Really destroying SIP dialog '07a1addc2d684b28a5257ea364141cb9' Method: ACK
  503. [Jun 21 15:22:44] DEBUG[1792]: rtp_engine.c:402 instance_destructor: Destroyed RTP instance '0x7f0f60008960'
  504. [Jun 21 15:22:50] DEBUG[1763]: cdr.c:4259 ast_cdr_engine_term: CDR Engine termination request received; waiting on messages...
  505. Asterisk cleanly ending (0).
  506. Executing last minute cleanups
  507. == Destroying musiconhold processes
  508. [Jun 21 15:22:50] DEBUG[1763]: res_musiconhold.c:1581 moh_class_destructor: Destroying MOH class 'default'
  509. == Manager unregistered action DBGet
  510. == Manager unregistered action DBPut
  511. == Manager unregistered action DBDel
  512. == Manager unregistered action DBDelTree
  513. [Jun 21 15:22:50] DEBUG[1763]: asterisk.c:2199 really_quit: Asterisk ending (0).
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