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  1. Asterisk 16.17.0, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.
  2. Created by Mark Spencer <markster@digium.com>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 16.17.0 currently running on freepbx (pid = 2462)
  9. Reliably Transmitting (no NAT) to 192.168.1.35:5060:
  10. OPTIONS sip:192.168.1.35 SIP/2.0
  11. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK1bfbf362
  12. Max-Forwards: 70
  13. From: "Unknown" <sip:Unknown@192.168.1.208:5160>;tag=as6811bc56
  14. To: <sip:192.168.1.35>
  15. Contact: <sip:Unknown@192.168.1.208:5160>
  16. Call-ID: 63d5fc1723ed2cd863e86c1438904de2@192.168.1.208:5160
  17. CSeq: 102 OPTIONS
  18. User-Agent: FPBX-15.0.17.34(16.17.0)
  19. Date: Sat, 27 Nov 2021 11:01:38 GMT
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  21. Supported: replaces, timer
  22. Content-Length: 0
  23.  
  24.  
  25. ---
  26.  
  27. <--- SIP read from UDP:192.168.1.35:5060 --->
  28. SIP/2.0 404 Not Found
  29. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK1bfbf362;received=192.168.1.208;rport=5160
  30. From: "Unknown" <sip:Unknown@192.168.1.208:5160>;tag=as6811bc56
  31. To: <sip:192.168.1.35>;tag=as085708e8
  32. Call-ID: 63d5fc1723ed2cd863e86c1438904de2@192.168.1.208:5160
  33. CSeq: 102 OPTIONS
  34. Server: VoxStack Wireless Gateway
  35. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  36. Supported: replaces, timer
  37. Accept: application/sdp
  38. Content-Length: 0
  39.  
  40. <------------->
  41. --- (11 headers 0 lines) ---
  42. Really destroying SIP dialog '63d5fc1723ed2cd863e86c1438904de2@192.168.1.208:5160' Method: OPTIONS
  43. == Using SIP RTP TOS bits 184
  44. == Using SIP RTP CoS mark 5
  45. Audio is at 13344
  46. Adding codec ulaw to SDP
  47. Adding codec alaw to SDP
  48. Adding non-codec 0x1 (telephone-event) to SDP
  49. Reliably Transmitting (no NAT) to 192.168.1.35:5060:
  50. INVITE sip:+91XxXxXxXxXx@192.168.1.35 SIP/2.0
  51. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121
  52. Max-Forwards: 70
  53. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  54. To: <sip:+91XxXxXxXxXx@192.168.1.35>
  55. Contact: <sip:anonymous@192.168.1.208:5160>
  56. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  57. CSeq: 102 INVITE
  58. User-Agent: FPBX-15.0.17.34(16.17.0)
  59. Date: Sat, 27 Nov 2021 11:01:43 GMT
  60. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  61. Supported: replaces, timer
  62. Content-Type: application/sdp
  63. Content-Length: 278
  64.  
  65. v=0
  66. o=root 1148483115 1148483115 IN IP4 192.168.1.208
  67. s=Asterisk PBX 16.17.0
  68. c=IN IP4 192.168.1.208
  69. t=0 0
  70. m=audio 13344 RTP/AVP 0 8 101
  71. a=rtpmap:0 PCMU/8000
  72. a=rtpmap:8 PCMA/8000
  73. a=rtpmap:101 telephone-event/8000
  74. a=fmtp:101 0-16
  75. a=ptime:20
  76. a=maxptime:150
  77. a=sendrecv
  78.  
  79. ---
  80. -- Called 9001/+91XxXxXxXxXx
  81.  
  82. <--- SIP read from UDP:192.168.1.35:5060 --->
  83. SIP/2.0 100 Trying
  84. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121;received=192.168.1.208;rport=5160
  85. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  86. To: <sip:+91XxXxXxXxXx@192.168.1.35>
  87. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  88. CSeq: 102 INVITE
  89. Server: VoxStack Wireless Gateway
  90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  91. Supported: replaces, timer
  92. Contact: <sip:+91XxXxXxXxXx@192.168.1.35:5060>
  93. Content-Length: 0
  94.  
  95. <------------->
  96. --- (11 headers 0 lines) ---
  97.  
  98. <--- SIP read from UDP:192.168.1.35:5060 --->
  99. SIP/2.0 100 Trying
  100. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121;received=192.168.1.208;rport=5160
  101. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  102. To: <sip:+91XxXxXxXxXx@192.168.1.35>
  103. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  104. CSeq: 102 INVITE
  105. Server: VoxStack Wireless Gateway
  106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  107. Supported: replaces, timer
  108. Contact: <sip:+91XxXxXxXxXx@192.168.1.35:5060>
  109. Content-Length: 0
  110.  
  111. <------------->
  112. --- (11 headers 0 lines) ---
  113.  
  114. <--- SIP read from UDP:192.168.1.35:5060 --->
  115. SIP/2.0 183 Session Progress
  116. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121;received=192.168.1.208;rport=5160
  117. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  118. To: <sip:+91XxXxXxXxXx@192.168.1.35>;tag=as77df4e58
  119. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  120. CSeq: 102 INVITE
  121. Server: VoxStack Wireless Gateway
  122. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  123. Supported: replaces, timer
  124. Contact: <sip:+91XxXxXxXxXx@192.168.1.35:5060>
  125. Content-Type: application/sdp
  126. Content-Length: 263
  127.  
  128. v=0
  129. o=root 250614339 250614339 IN IP4 192.168.1.35
  130. s=VoxStack Wireless Gateway
  131. c=IN IP4 192.168.1.35
  132. t=0 0
  133. m=audio 13760 RTP/AVP 0 8 101
  134. a=rtpmap:0 PCMU/8000
  135. a=rtpmap:8 PCMA/8000
  136. a=rtpmap:101 telephone-event/8000
  137. a=fmtp:101 0-16
  138. a=ptime:20
  139. a=sendrecv
  140. <------------->
  141. --- (12 headers 12 lines) ---
  142. sip_route_dump: route/path hop: <sip:+91XxXxXxXxXx@192.168.1.35:5060>
  143. Got SDP version 250614339 and unique parts [root 250614339 IN IP4 192.168.1.35]
  144. Found RTP audio format 0
  145. Found RTP audio format 8
  146. Found RTP audio format 101
  147. Found audio description format PCMU for ID 0
  148. Found audio description format PCMA for ID 8
  149. Found audio description format telephone-event for ID 101
  150. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  151. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  152. > 0x7f4c080418b0 -- Strict RTP learning after remote address set to: 192.168.1.35:13760
  153. Peer audio RTP is at port 192.168.1.35:13760
  154. -- SIP/9001-00000008 is making progress
  155. > 0x7f4c080418b0 -- Strict RTP switching to RTP target address 192.168.1.35:13760 as source
  156. > 0x7f4c080418b0 -- Strict RTP learning complete - Locking on source address 192.168.1.35:13760
  157. Scheduling destruction of SIP dialog '2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35' in 6400 ms (Method: INVITE)
  158. Reliably Transmitting (no NAT) to 192.168.1.35:5060:
  159. CANCEL sip:+91XxXxXxXxXx@192.168.1.35 SIP/2.0
  160. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121
  161. Max-Forwards: 70
  162. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  163. To: <sip:+91XxXxXxXxXx@192.168.1.35>
  164. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  165. CSeq: 102 CANCEL
  166. User-Agent: FPBX-15.0.17.34(16.17.0)
  167. Content-Length: 0
  168.  
  169.  
  170. ---
  171. Scheduling destruction of SIP dialog '2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35' in 6400 ms (Method: INVITE)
  172.  
  173. <--- SIP read from UDP:192.168.1.35:5060 --->
  174. SIP/2.0 487 Request Terminated
  175. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121;received=192.168.1.208;rport=5160
  176. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  177. To: <sip:+91XxXxXxXxXx@192.168.1.35>;tag=as77df4e58
  178. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  179. CSeq: 102 INVITE
  180. Server: VoxStack Wireless Gateway
  181. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  182. Supported: replaces, timer
  183. Content-Length: 0
  184.  
  185. <------------->
  186. --- (10 headers 0 lines) ---
  187. Transmitting (no NAT) to 192.168.1.35:5060:
  188. ACK sip:+91XxXxXxXxXx@192.168.1.35:5060 SIP/2.0
  189. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121
  190. Max-Forwards: 70
  191. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  192. To: <sip:+91XxXxXxXxXx@192.168.1.35>;tag=as77df4e58
  193. Contact: <sip:anonymous@192.168.1.208:5160>
  194. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  195. CSeq: 102 ACK
  196. User-Agent: FPBX-15.0.17.34(16.17.0)
  197. Content-Length: 0
  198.  
  199.  
  200. ---
  201. Scheduling destruction of SIP dialog '2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35' in 6400 ms (Method: INVITE)
  202.  
  203. <--- SIP read from UDP:192.168.1.35:5060 --->
  204. SIP/2.0 200 OK
  205. Via: SIP/2.0/UDP 192.168.1.208:5160;branch=z9hG4bK5cb8e121;received=192.168.1.208;rport=5160
  206. From: "Anonymous" <sip:anonymous@192.168.1.35:5160>;tag=as00c86d39
  207. To: <sip:+91XxXxXxXxXx@192.168.1.35>;tag=as77df4e58
  208. Call-ID: 2902a49556e2b87f7938cf0f3fe4ccf6@192.168.1.35
  209. CSeq: 102 CANCEL
  210. Server: VoxStack Wireless Gateway
  211. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  212. Supported: replaces, timer
  213. Content-Length: 0
  214.  
  215. <------------->
  216. --- (10 headers 0 lines) ---
  217.  
  218.  
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