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Jun 16th, 2017
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  1. Transmitting (NAT) to xxx.xxx.xxx.xxx:38148:
  2. SIP/2.0 180 Ringing
  3. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  4. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  5. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  6. Call-ID: 581148542@192_168_102_10
  7. CSeq: 3 INVITE
  8. User-Agent: Asterisk PBX
  9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  10. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  11. Content-Length: 0
  12.  
  13.  
  14. ---
  15. We're at yyy.yyy.yyy.yyy port 19832
  16. Adding codec 0x4 (ulaw) to SDP
  17. Adding codec 0x8 (alaw) to SDP
  18. Adding non-codec 0x1 (telephone-event) to SDP
  19. Transmitting (NAT) to xxx.xxx.xxx.xxx:38148:
  20. SIP/2.0 183 Session Progress
  21. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  22. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  23. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  24. Call-ID: 581148542@192_168_102_10
  25. CSeq: 3 INVITE
  26. User-Agent: Asterisk PBX
  27. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  28. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  29. Content-Type: application/sdp
  30. Content-Length: 244
  31.  
  32. v=0
  33. o=root 14414 14414 IN IP4 yyy.yyy.yyy.yyy
  34. s=session
  35. c=IN IP4 yyy.yyy.yyy.yyy
  36. t=0 0
  37. m=audio 19832 RTP/AVP 0 8 101
  38. a=rtpmap:0 PCMU/8000
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:101 telephone-event/8000
  41. a=fmtp:101 0-16
  42. a=silenceSupp:off - - - -
  43.  
  44. ---
  45. We're at yyy.yyy.yyy.yyy port 19832
  46. Adding codec 0x4 (ulaw) to SDP
  47. Adding codec 0x8 (alaw) to SDP
  48. Adding non-codec 0x1 (telephone-event) to SDP
  49. Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:38148:
  50. SIP/2.0 200 OK
  51. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  52. From: "Joe Bloggst" <sip:John-office@asterisk.domain.net>;tag=3735246692
  53. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  54. Call-ID: 581148542@192_168_102_10
  55. CSeq: 3 INVITE
  56. User-Agent: Asterisk PBX
  57. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  58. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  59. Content-Type: application/sdp
  60. Content-Length: 244
  61.  
  62. v=0
  63. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  64. s=session
  65. c=IN IP4 yyy.yyy.yyy.yyy
  66. t=0 0
  67. m=audio 19832 RTP/AVP 0 8 101
  68. a=rtpmap:0 PCMU/8000
  69. a=rtpmap:8 PCMA/8000
  70. a=rtpmap:101 telephone-event/8000
  71. a=fmtp:101 0-16
  72. a=silenceSupp:off - - - -
  73.  
  74. ---
  75. Retransmitting #1 (NAT) to xxx.xxx.xxx.xxx:38148:
  76. SIP/2.0 200 OK
  77. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  78. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  79. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  80. Call-ID: 581148542@192_168_102_10
  81. CSeq: 3 INVITE
  82. User-Agent: Asterisk PBX
  83. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  84. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  85. Content-Type: application/sdp
  86. Content-Length: 244
  87.  
  88. v=0
  89. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  90. s=session
  91. c=IN IP4 yyy.yyy.yyy.yyy
  92. t=0 0
  93. m=audio 19832 RTP/AVP 0 8 101
  94. a=rtpmap:0 PCMU/8000
  95. a=rtpmap:8 PCMA/8000
  96. a=rtpmap:101 telephone-event/8000
  97. a=fmtp:101 0-16
  98. a=silenceSupp:off - - - -
  99.  
  100. ---
  101. Retransmitting #2 (NAT) to xxx.xxx.xxx.xxx:38148:
  102. SIP/2.0 200 OK
  103. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  104. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  105. To: <sip:0016319240517@asterisk.domains.net>;user=phone;tag=as43e72b43
  106. Call-ID: 581148542@192_168_102_10
  107. CSeq: 3 INVITE
  108. User-Agent: Asterisk PBX
  109. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  110. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  111. Content-Type: application/sdp
  112. Content-Length: 244
  113.  
  114. v=0
  115. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  116. s=session
  117. c=IN IP4 yyy.yyy.yyy.yyy
  118. t=0 0
  119. m=audio 19832 RTP/AVP 0 8 101
  120. a=rtpmap:0 PCMU/8000
  121. a=rtpmap:8 PCMA/8000
  122. a=rtpmap:101 telephone-event/8000
  123. a=fmtp:101 0-16
  124. a=silenceSupp:off - - - -
  125.  
  126. ---
  127.  
  128. <-- SIP read from xxx.xxx.xxx.xxx:38148:
  129.  
  130. --- (0 headers 0 lines) Nat keepalive ---
  131. Retransmitting #3 (NAT) to xxx.xxx.xxx.xxx:38148:
  132. SIP/2.0 200 OK
  133. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  134. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  135. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  136. Call-ID: 581148542@192_168_102_10
  137. CSeq: 3 INVITE
  138. User-Agent: Asterisk PBX
  139. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  140. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  141. Content-Type: application/sdp
  142. Content-Length: 244
  143.  
  144. v=0
  145. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  146. s=session
  147. c=IN IP4 yyy.yyy.yyy.yyy
  148. t=0 0
  149. m=audio 19832 RTP/AVP 0 8 101
  150. a=rtpmap:0 PCMU/8000
  151. a=rtpmap:8 PCMA/8000
  152. a=rtpmap:101 telephone-event/8000
  153. a=fmtp:101 0-16
  154. a=silenceSupp:off - - - -
  155.  
  156. ---
  157. Retransmitting #4 (NAT) to xxx.xxx.xxx.xxx:38148:
  158. SIP/2.0 200 OK
  159. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  160. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  161. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  162. Call-ID: 581148542@192_168_102_10
  163. CSeq: 3 INVITE
  164. User-Agent: Asterisk PBX
  165. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  166. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  167. Content-Type: application/sdp
  168. Content-Length: 244
  169.  
  170. v=0
  171. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  172. s=session
  173. c=IN IP4 yyy.yyy.yyy.yyy
  174. t=0 0
  175. m=audio 19832 RTP/AVP 0 8 101
  176. a=rtpmap:0 PCMU/8000
  177. a=rtpmap:8 PCMA/8000
  178. a=rtpmap:101 telephone-event/8000
  179. a=fmtp:101 0-16
  180. a=silenceSupp:off - - - -
  181.  
  182. ---
  183. Retransmitting #5 (NAT) to xxx.xxx.xxx.xxx:38148:
  184. SIP/2.0 200 OK
  185. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  186. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  187. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  188. Call-ID: 581148542@192_168_102_10
  189. CSeq: 3 INVITE
  190. User-Agent: Asterisk PBX
  191. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  192. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  193. Content-Type: application/sdp
  194. Content-Length: 244
  195.  
  196. v=0
  197. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  198. s=session
  199. c=IN IP4 yyy.yyy.yyy.yyy
  200. t=0 0
  201. m=audio 19832 RTP/AVP 0 8 101
  202. a=rtpmap:0 PCMU/8000
  203. a=rtpmap:8 PCMA/8000
  204. a=rtpmap:101 telephone-event/8000
  205. a=fmtp:101 0-16
  206. a=silenceSupp:off - - - -
  207.  
  208. ---
  209. Retransmitting #6 (NAT) to xxx.xxx.xxx.xxx:38148:
  210. SIP/2.0 200 OK
  211. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  212. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  213. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as43e72b43
  214. Call-ID: 581148542@192_168_102_10
  215. CSeq: 3 INVITE
  216. User-Agent: Asterisk PBX
  217. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  218. Contact: <sip:0016319240517@yyy.yyy.yyy.yyy>
  219. Content-Type: application/sdp
  220. Content-Length: 244
  221.  
  222. v=0
  223. o=root 14414 14415 IN IP4 yyy.yyy.yyy.yyy
  224. s=session
  225. c=IN IP4 yyy.yyy.yyy.yyy
  226. t=0 0
  227. m=audio 19832 RTP/AVP 0 8 101
  228. a=rtpmap:0 PCMU/8000
  229. a=rtpmap:8 PCMA/8000
  230. a=rtpmap:101 telephone-event/8000
  231. a=fmtp:101 0-16
  232. a=silenceSupp:off - - - -
  233.  
  234. ---
  235. Sep 23 13:05:32 WARNING[14424]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 581148542@192_168_102_10 for seqno 3 (Critical Response)
  236. Sep 23 13:05:32 WARNING[14424]: chan_sip.c:1243 retrans_pkt: Hanging up call 581148542@192_168_102_10 - no reply to our critical packet.
  237. Destroying call '581148542@192_168_102_10'
  238.  
  239. <-- SIP read from xxx.xxx.xxx.xxx:38148:
  240.  
  241. --- (0 headers 0 lines) Nat keepalive ---
  242.  
  243. <-- SIP read from xxx.xxx.xxx.xxx:38148:
  244. CANCEL sip:0016319240517@asterisk.domain.net SIP/2.0
  245. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;rport
  246. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  247. To: <sip:0016319240517@asterisk.domain.net>;user=phone
  248. Call-ID: 581148542@192_168_102_10
  249. CSeq: 3 CANCEL
  250. Contact: <sip:John-office@xxx.xxx.xxx.xxx:5060>
  251. Proxy-Authorization: Digest username="John-office", realm="asterisk", algorithm=MD5, uri="sip:0016319240517@asterisk.domain.net", nonce="096f4c45", response="bce8071787e9f92a7cab6a677c321e9f"
  252. Max-Forwards: 70
  253. User-Agent: A580 IP021920000000
  254. Content-Length: 0
  255.  
  256.  
  257. --- (11 headers 0 lines) ---
  258. Transmitting (no NAT) to xxx.xxx.xxx.xxx:38148:
  259. SIP/2.0 481 Call leg/transaction does not exist
  260. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK284275c77402e0b1be37c6f28762c61a;received=xxx.xxx.xxx.xxx;rport=38148
  261. From: "Joe Bloggs" <sip:John-office@asterisk.domain.net>;tag=3735246692
  262. To: <sip:0016319240517@asterisk.domain.net>;user=phone;tag=as5214a1c6
  263. Call-ID: 581148542@192_168_102_10
  264. CSeq: 3 CANCEL
  265. User-Agent: Asterisk PBX
  266. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  267. Content-Length: 0
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