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  1. Authorization: Digest username="2001", realm="asterisk", nonce="0a5e8b73", uri="                                                                                        sip:2001@192.168.1.201", response="1d34a23f5f021d4a41cc9cb74ec5a085", algorithm=                                                                                        MD5
  2. Content-Length: 0
  3.  
  4. <------------->
  5. --- (12 headers 0 lines) ---
  6. Sending to 192.168.1.4:5070 (NAT)
  7. Reliably Transmitting (NAT) to 192.168.1.4:5070:
  8. OPTIONS sip:2001@192.168.1.4:5070 SIP/2.0
  9. Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK6ec5a74d;rport
  10. Max-Forwards: 70
  11. From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as107190b6
  12. To: <sip:2001@192.168.1.4:5070>
  13. Contact: <sip:asterisk@192.168.1.201:5060>
  14. Call-ID: 45052561509faaae7b88e88a5e696584@192.168.1.201:5060
  15. CSeq: 102 OPTIONS
  16. User-Agent: Asterisk PBX 1.8.21.0
  17. Date: Sun, 28 Apr 2013 18:35:47 GMT
  18. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
  19. Supported: replaces, timer
  20. Content-Length: 0
  21.  
  22.  
  23. ---
  24.  
  25. <--- Transmitting (NAT) to 192.168.1.4:5070 --->
  26. SIP/2.0 200 OK
  27. Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj7f4b697d16424f93947d36a15a7ef5                                                                                        e4;received=192.168.1.4;rport=5070
  28. From: <sip:2001@192.168.1.201>;tag=c624379ced5249b2a645a752167dbb85
  29. To: <sip:2001@192.168.1.201>;tag=as43a84e39
  30. Call-ID: 802c20bc482943299346cb4130d258d6
  31. CSeq: 18220 REGISTER
  32. Server: Asterisk PBX 1.8.21.0
  33. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
  34. Supported: replaces, timer
  35. Expires: 300
  36. Contact: <sip:2001@192.168.1.4:5070>;expires=300
  37. Date: Sun, 28 Apr 2013 18:35:47 GMT
  38. Content-Length: 0
  39.  
  40.  
  41. <------------>
  42. Scheduling destruction of SIP dialog '10338ff02e6f968b07fb71e357aed3d9@192.168.1                                                                                        .201:5060' in 6400 ms (Method: NOTIFY)
  43. Reliably Transmitting (NAT) to 192.168.1.4:5070:
  44. NOTIFY sip:2001@192.168.1.4:5070 SIP/2.0
  45. Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1302d636;rport
  46. Max-Forwards: 70
  47. From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as3469d14e
  48. To: <sip:2001@192.168.1.4:5070>
  49. Contact: <sip:asterisk@192.168.1.201:5060>
  50. Call-ID: 10338ff02e6f968b07fb71e357aed3d9@192.168.1.201:5060
  51. CSeq: 102 NOTIFY
  52. User-Agent: Asterisk PBX 1.8.21.0
  53. Event: message-summary
  54. Content-Type: application/simple-message-summary
  55. Content-Length: 93
  56.  
  57. Messages-Waiting: no
  58. Message-Account: sip:asterisk@192.168.1.201
  59. Voice-Message: 0/0 (0/0)
  60.  
  61. ---
  62. Scheduling destruction of SIP dialog '802c20bc482943299346cb4130d258d6' in 32000                                                                                         ms (Method: REGISTER)
  63.  
  64. <--- SIP read from UDP:192.168.1.4:5070 --->
  65. SIP/2.0 200 OK
  66. Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.201;branch=z9h                                                                                        G4bK6ec5a74d
  67. Call-ID: 45052561509faaae7b88e88a5e696584@192.168.1.201:5060
  68. From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as107190b6
  69. To: <sip:2001@192.168.1.4>
  70. CSeq: 102 OPTIONS
  71. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAG                                                                                        E, OPTIONS
  72. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/si                                                                                        pfrag;version=2.0, application/im-iscomposing+xml, text/plain
  73. Supported: 100rel, norefersub
  74. Allow-Events: presence, refer
  75. User-Agent: AdoreSoftphone
  76. Content-Type: application/sdp
  77. Content-Length: 718
  78.  
  79. v=0
  80. o=- 3576176452 3576176452 IN IP4 192.168.1.4
  81. s=pjmedia
  82. c=IN IP4 192.168.1.4
  83. t=0 0
  84. m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
  85. a=rtcp:4001 IN IP4 192.168.1.4
  86. a=rtpmap:103 speex/16000
  87. a=rtpmap:102 speex/8000
  88. a=rtpmap:104 speex/32000
  89. a=rtpmap:117 iLBC/8000
  90. a=fmtp:117 mode=30
  91. a=rtpmap:3 GSM/8000
  92. a=rtpmap:0 PCMU/8000
  93. a=rtpmap:8 PCMA/8000
  94. a=rtpmap:9 G722/8000
  95. a=rtpmap:118 AMR/8000
  96. a=rtpmap:119 AMR-WB/16000
  97. a=rtpmap:18 G729/8000
  98. a=rtpmap:4 G723/8000
  99. a=rtpmap:2 G726-32/8000
  100. a=rtpmap:15 G728/8000
  101. a=rtpmap:125 G7221/16000
  102. a=fmtp:125 bitrate=24000
  103. a=rtpmap:126 G7221/16000
  104. a=fmtp:126 bitrate=32000
  105. a=sendrecv
  106. a=rtpmap:101 telephone-event/8000
  107. a=fmtp:101 0-15
  108. <------------->
  109. --- (13 headers 29 lines) ---
  110. Really destroying SIP dialog '45052561509faaae7b88e88a5e696584@192.168.1.201:506                                                                                        0' Method: OPTIONS
  111.  
  112. <--- SIP read from UDP:192.168.1.4:5070 --->
  113. SIP/2.0 200 OK
  114. Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.201;branch=z9h                                                                                        G4bK1302d636
  115. Call-ID: 10338ff02e6f968b07fb71e357aed3d9@192.168.1.201:5060
  116. From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as3469d14e
  117. To: <sip:2001@192.168.1.4>
  118. CSeq: 102 NOTIFY
  119. Content-Length: 0
  120.  
  121. <------------->
  122. --- (7 headers 0 lines) ---
  123. Really destroying SIP dialog '10338ff02e6f968b07fb71e357aed3d9@192.168.1.201:506                                                                                        0' Method: NOTIFY
  124.  
  125. <--- SIP read from UDP:192.168.1.202:5060 --->
  126. INVITE sip:2001@192.168.1.201 SIP/2.0
  127. Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7321dd16;rport
  128. Max-Forwards: 70
  129. From: "3000" <sip:3000@192.168.1.202>;tag=as146d1bbf
  130. To: <sip:2001@192.168.1.201>
  131. Contact: <sip:3000@192.168.1.202:5060>
  132. Call-ID: 063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060
  133. CSeq: 102 INVITE
  134. User-Agent: Asterisk PBX 1.8.21.0
  135. Date: Mon, 29 Apr 2013 01:21:03 GMT
  136. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
  137. Supported: replaces, timer
  138. Content-Type: application/sdp
  139. Content-Length: 260
  140.  
  141. v=0
  142. o=root 205266058 205266058 IN IP4 192.168.1.202
  143. s=Asterisk PBX 1.8.21.0
  144. c=IN IP4 192.168.1.202
  145. t=0 0
  146. m=audio 16618 RTP/AVP 0 3 101
  147. a=rtpmap:0 PCMU/8000
  148. a=rtpmap:3 GSM/8000
  149. a=rtpmap:101 telephone-event/8000
  150. a=fmtp:101 0-16
  151. a=ptime:20
  152. a=sendrecv
  153. <------------->
  154. --- (14 headers 12 lines) ---
  155. Sending to 192.168.1.202:5060 (NAT)
  156. Using INVITE request as basis request - 063d0bb833d9a848487cbc944ddc02ae@192.168                                                                                        .1.202:5060
  157. Found peer 'central2' for '3000' from 192.168.1.202:5060
  158.  
  159. <--- Reliably Transmitting (NAT) to 192.168.1.202:5060 --->
  160. SIP/2.0 401 Unauthorized
  161. Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7321dd16;received=192.168.1.20                                                                                        2;rport=5060
  162. From: "3000" <sip:3000@192.168.1.202>;tag=as146d1bbf
  163. To: <sip:2001@192.168.1.201>;tag=as320d1474
  164. Call-ID: 063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060
  165. CSeq: 102 INVITE
  166. Server: Asterisk PBX 1.8.21.0
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
  168. Supported: replaces, timer
  169. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d877a7b"
  170. Content-Length: 0
  171.  
  172.  
  173. <------------>
  174. Scheduling destruction of SIP dialog '063d0bb833d9a848487cbc944ddc02ae@192.168.1                                                                                        .202:5060' in 6400 ms (Method: INVITE)
  175.  
  176. <--- SIP read from UDP:192.168.1.202:5060 --->
  177. ACK sip:2001@192.168.1.201 SIP/2.0
  178. Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK7321dd16;rport
  179. Max-Forwards: 70
  180. From: "3000" <sip:3000@192.168.1.202>;tag=as146d1bbf
  181. To: <sip:2001@192.168.1.201>;tag=as320d1474
  182. Contact: <sip:3000@192.168.1.202:5060>
  183. Call-ID: 063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:5060
  184. CSeq: 102 ACK
  185. User-Agent: Asterisk PBX 1.8.21.0
  186. Content-Length: 0
  187.  
  188. <------------->
  189. --- (10 headers 0 lines) ---
  190. Really destroying SIP dialog '063d0bb833d9a848487cbc944ddc02ae@192.168.1.202:506                                                                                        0' Method: ACK
  191.  
  192. <--- SIP read from UDP:192.168.1.4:5070 --->
  193.  
  194. <------------->
  195. Really destroying SIP dialog '0dcdd5c90953cab90f66f9d9678bfd31@192.168.1.202:506                                                                                        0' Method: OPTIONS
  196. home*CLI> sip set debug off
  197. SIP Debugging Disabled
  198. home*CLI> exi
  199. No such command 'exi' (type 'core show help exi' for other possible commands)
  200. home*CLI> exit
  201. Executing last minute cleanups
  202. Asterisk ending (0).
  203. root@home:/etc/asterisk# nano debugc1.txt
  204.   GNU nano 2.2.4                                         Fichero: debugc1.txt
  205.  
  206. Parsing /etc/asterisk/asterisk.conf
  207. Seeding global EID '00:01:6c:c7:7f:a3' from 'eth0' using 'siocgifhwaddr'
  208. Asterisk 1.8.21.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  209. Created by Mark Spencer <markster@digium.com>
  210. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  211. This is free software, with components licensed under the GNU General Public
  212. License version 2 and other licenses; you are welcome to redistribute it under
  213. certain conditions. Type 'core show license' for details.
  214. =========================================================================
  215. Connected to Asterisk 1.8.21.0 currently running on home (pid = 1829)
  216. home*CLI> ^M^[[0KVerbosity is at least 53
  217. ^M^[[Khome*CLI> ^M^[[0KCore debug is at least 33
  218. ^M^[[Khome*CLI> exit^H^H^H^Hsip set debug off^H^Hn^[[K
  219. home*CLI> ^M^[[0KSIP Debugging enabled
  220. ^M^[[Khome*CLI> ^M^[[0KReliably Transmitting (no NAT) to 200.43.153.237:5060:
  221. OPTIONS sip:1000@200.43.153.237 SIP/2.0
  222. Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1df334e4
  223. Max-Forwards: 70
  224. From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as1b88b214
  225. To: <sip:1000@200.43.153.237>
  226. Contact: <sip:asterisk@192.168.1.201:5060>
  227. Call-ID: 482cf3b05c4380d874b590a3463cec3f@192.168.1.201:5060
  228. CSeq: 102 OPTIONS
  229. User-Agent: Asterisk PBX 1.8.21.0
  230. Date: Sun, 28 Apr 2013 18:35:39 GMT
  231. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  232. Supported: replaces, timer
  233. Content-Length: 0
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