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  1. [Nov 17 07:01:45] Retransmitting #2 (NAT) to 61.16.251.219:5060:
  2. OPTIONS sip:61.16.251.219 SIP/2.0
  3. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK4d23df24;rport
  4. From: "asterisk" <sip:[email protected]>;tag=as45d353d0
  5. To: <sip:61.16.251.219>
  6. Contact: <sip:[email protected]>
  7. Call-ID: [email protected]
  8. CSeq: 102 OPTIONS
  9. User-Agent: Asterisk PBX
  10. Max-Forwards: 70
  11. Date: Wed, 17 Nov 2010 13:01:43 GMT
  12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  13. Supported: replaces
  14. Content-Length: 0
  15.  
  16.  
  17. ---
  18. [Nov 17 07:01:45] == Parsing '/etc/asterisk/manager.conf': [Nov 17 07:01:45] Found
  19. [Nov 17 07:01:45] == Manager 'sendcron' logged on from 71.52.46.89
  20. [Nov 17 07:01:45] -- Executing [919782232254@default:1] AGI("Local/919782232254@default-2527,2", "agi://127.0.0.1:4577/call_log") in new stack
  21. [Nov 17 07:01:45] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
  22. [Nov 17 07:01:45] -- Executing [919782232254@default:2] Dial("Local/919782232254@default-2527,2", "SIP/919782232254@voipprovider|20|tTo") in new stack
  23. [Nov 17 07:01:45] Audio is at 71.52.46.89 port 15078
  24. [Nov 17 07:01:45] Adding codec 0x100 (g729) to SDP
  25. [Nov 17 07:01:45] Adding non-codec 0x1 (telephone-event) to SDP
  26. [Nov 17 07:01:45] Reliably Transmitting (NAT) to 64.56.64.43:5060:
  27. INVITE sip:[email protected] SIP/2.0
  28. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK21aaf05c;rport
  29. From: "V1170701450000015155" <sip:[email protected]>;tag=as05c4e844
  30. To: <sip:[email protected]>
  31. Contact: <sip:[email protected]>
  32. Call-ID: [email protected]
  33. CSeq: 102 INVITE
  34. User-Agent: Asterisk PBX
  35. Max-Forwards: 70
  36. Remote-Party-ID: "V1170701450000015155" <sip:[email protected]>;privacy=off;screen=no
  37. Date: Wed, 17 Nov 2010 13:01:45 GMT
  38. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  39. Supported: replaces
  40. Content-Type: application/sdp
  41. Content-Length: 259
  42.  
  43. v=0
  44. o=root 3599 3599 IN IP4 71.52.46.89
  45. s=session
  46. c=IN IP4 71.52.46.89
  47. t=0 0
  48. m=audio 15078 RTP/AVP 18 101
  49. a=rtpmap:18 G729/8000
  50. a=fmtp:18 annexb=no
  51. a=rtpmap:101 telephone-event/8000
  52. a=fmtp:101 0-16
  53. a=silenceSupp:off - - - -
  54. a=ptime:20
  55. a=sendrecv
  56.  
  57. ---
  58. [Nov 17 07:01:45] -- Called 919782232254@voipprovider
  59. [Nov 17 07:01:45]
  60. <--- SIP read from 64.56.64.43:5060 --->
  61. SIP/2.0 100 Trying
  62. CSeq: 102 INVITE
  63. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK21aaf05c
  64. From: "V1170701450000015155" <sip:[email protected]>;tag=as05c4e844
  65. Call-ID: [email protected]
  66. To: <sip:[email protected]>;tag=171101100546201580157837
  67. Contact: <sip:64.56.64.43:5060;transport=udp>
  68. Content-Length: 0
  69.  
  70.  
  71. <------------->
  72. [Nov 17 07:01:45] --- (8 headers 0 lines) ---
  73. [Nov 17 07:01:45] == Parsing '/etc/asterisk/manager.conf': [Nov 17 07:01:45] Found
  74. [Nov 17 07:01:45] == Manager 'sendcron' logged on from 71.52.46.89
  75. [Nov 17 07:01:45] -- Executing [919792232255@default:1] AGI("Local/919792232255@default-1a60,2", "agi://127.0.0.1:4577/call_log") in new stack
  76. [Nov 17 07:01:45] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
  77. [Nov 17 07:01:45] -- Executing [919792232255@default:2] Dial("Local/919792232255@default-1a60,2", "SIP/919792232255@voipprovider|20|tTo") in new stack
  78. [Nov 17 07:01:45] Audio is at 71.52.46.89 port 16512
  79. [Nov 17 07:01:45] Adding codec 0x100 (g729) to SDP
  80. [Nov 17 07:01:45] Adding non-codec 0x1 (telephone-event) to SDP
  81. [Nov 17 07:01:45] Reliably Transmitting (NAT) to 64.56.64.43:5060:
  82. INVITE sip:[email protected] SIP/2.0
  83. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK03cb7a84;rport
  84. From: "V1170701450000015156" <sip:[email protected]>;tag=as7890cf45
  85. To: <sip:[email protected]>
  86. Contact: <sip:[email protected]>
  87. Call-ID: [email protected]
  88. CSeq: 102 INVITE
  89. User-Agent: Asterisk PBX
  90. Max-Forwards: 70
  91. Remote-Party-ID: "V1170701450000015156" <sip:[email protected]>;privacy=off;screen=no
  92. Date: Wed, 17 Nov 2010 13:01:45 GMT
  93. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  94. Supported: replaces
  95. Content-Type: application/sdp
  96. Content-Length: 259
  97.  
  98. v=0
  99. o=root 3599 3599 IN IP4 71.52.46.89
  100. s=session
  101. c=IN IP4 71.52.46.89
  102. t=0 0
  103. m=audio 16512 RTP/AVP 18 101
  104. a=rtpmap:18 G729/8000
  105. a=fmtp:18 annexb=no
  106. a=rtpmap:101 telephone-event/8000
  107. a=fmtp:101 0-16
  108. a=silenceSupp:off - - - -
  109. a=ptime:20
  110. a=sendrecv
  111.  
  112. ---
  113. [Nov 17 07:01:45] -- Called 919792232255@voipprovider
  114. [Nov 17 07:01:45]
  115. <--- SIP read from 64.56.64.43:5060 --->
  116. SIP/2.0 100 Trying
  117. CSeq: 102 INVITE
  118. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK03cb7a84
  119. From: "V1170701450000015156" <sip:[email protected]>;tag=as7890cf45
  120. Call-ID: [email protected]
  121. To: <sip:[email protected]>;tag=171101100546201581407845
  122. Contact: <sip:64.56.64.43:5060;transport=udp>
  123. Content-Length: 0
  124.  
  125.  
  126. <------------->
  127. [Nov 17 07:01:45] --- (8 headers 0 lines) ---
  128. [Nov 17 07:01:45]
  129. <--- SIP read from 64.56.64.43:5060 --->
  130. SIP/2.0 183 Session Progress
  131. CSeq: 102 INVITE
  132. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK03cb7a84
  133. From: "V1170701450000015156" <sip:[email protected]>;tag=as7890cf45
  134. Call-ID: [email protected]
  135. To: <sip:[email protected]>;tag=171101100546201581407845
  136. Contact: <sip:64.56.64.43:5060;transport=udp>
  137. Content-Type: application/sdp
  138. Content-Length: 244
  139.  
  140. v=0
  141. o=VoipSwitch 8844 8844 IN IP4 64.56.64.43
  142. s=VoipSIP
  143. i=Audio Session
  144. c=IN IP4 64.56.64.43
  145. t=0 0
  146. m=audio 7844 RTP/AVP 18 101
  147. a=rtpmap:18 G729/8000/1
  148. a=fmtp:18 annexb=no
  149. a=rtpmap:101 telephone-event/8000
  150. a=fmtp:101 0-15
  151. a=sendrecv
  152.  
  153. <------------->
  154. [Nov 17 07:01:45] --- (9 headers 12 lines) ---
  155. [Nov 17 07:01:45] Found RTP audio format 18
  156. [Nov 17 07:01:45] Found RTP audio format 101
  157. [Nov 17 07:01:45] Peer audio RTP is at port 64.56.64.43:7844
  158. [Nov 17 07:01:45] Found audio description format G729 for ID 18
  159. [Nov 17 07:01:45] Found audio description format telephone-event for ID 101
  160. [Nov 17 07:01:45] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
  161. [Nov 17 07:01:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  162. [Nov 17 07:01:45] Peer audio RTP is at port 64.56.64.43:7844
  163. [Nov 17 07:01:45] -- SIP/voipprovider-1321cc70 is making progress passing it to Local/919792232255@default-1a60,2
  164. [Nov 17 07:01:46] Retransmitting #3 (NAT) to 61.16.251.219:5060:
  165. OPTIONS sip:61.16.251.219 SIP/2.0
  166. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK4d23df24;rport
  167. From: "asterisk" <sip:[email protected]>;tag=as45d353d0
  168. To: <sip:61.16.251.219>
  169. Contact: <sip:[email protected]>
  170. Call-ID: [email protected]
  171. CSeq: 102 OPTIONS
  172. User-Agent: Asterisk PBX
  173. Max-Forwards: 70
  174. Date: Wed, 17 Nov 2010 13:01:43 GMT
  175. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  176. Supported: replaces
  177. Content-Length: 0
  178.  
  179.  
  180. ---
  181. [Nov 17 07:01:46]
  182. <--- SIP read from 64.56.64.43:5060 --->
  183. SIP/2.0 183 Session Progress
  184. CSeq: 102 INVITE
  185. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK21aaf05c
  186. From: "V1170701450000015155" <sip:[email protected]>;tag=as05c4e844
  187. Call-ID: [email protected]
  188. To: <sip:[email protected]>;tag=171101100546201580157837
  189. Contact: <sip:64.56.64.43:5060;transport=udp>
  190. Content-Type: application/sdp
  191. Content-Length: 244
  192.  
  193. v=0
  194. o=VoipSwitch 8836 8836 IN IP4 64.56.64.43
  195. s=VoipSIP
  196. i=Audio Session
  197. c=IN IP4 64.56.64.43
  198. t=0 0
  199. m=audio 7836 RTP/AVP 18 101
  200. a=rtpmap:18 G729/8000/1
  201. a=fmtp:18 annexb=no
  202. a=rtpmap:101 telephone-event/8000
  203. a=fmtp:101 0-15
  204. a=sendrecv
  205.  
  206. <------------->
  207. [Nov 17 07:01:46] --- (9 headers 12 lines) ---
  208. [Nov 17 07:01:46] Found RTP audio format 18
  209. [Nov 17 07:01:46] Found RTP audio format 101
  210. [Nov 17 07:01:46] Peer audio RTP is at port 64.56.64.43:7836
  211. [Nov 17 07:01:46] Found audio description format G729 for ID 18
  212. [Nov 17 07:01:46] Found audio description format telephone-event for ID 101
  213. [Nov 17 07:01:46] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
  214. [Nov 17 07:01:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  215. [Nov 17 07:01:46] Peer audio RTP is at port 64.56.64.43:7836
  216. [Nov 17 07:01:46] -- SIP/voipprovider-1321f6a0 is making progress passing it to Local/919782232254@default-2527,2
  217. [Nov 17 07:01:47]
  218. <--- SIP read from 115.242.50.33:55598 --->
  219.  
  220.  
  221.  
  222. <------------->
  223. [Nov 17 07:01:47] Retransmitting #4 (NAT) to 61.16.251.219:5060:
  224. OPTIONS sip:61.16.251.219 SIP/2.0
  225. Via: SIP/2.0/UDP 71.52.46.89:5060;branch=z9hG4bK4d23df24;rport
  226. From: "asterisk" <sip:[email protected]>;tag=as45d353d0
  227. To: <sip:61.16.251.219>
  228. Contact: <sip:[email protected]>
  229. Call-ID: [email protected]
  230. CSeq: 102 OPTIONS
  231. User-Agent: Asterisk PBX
  232. Max-Forwards: 70
  233. Date: Wed, 17 Nov 2010 13:01:43 GMT
  234. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  235. Supported: replaces
  236. Content-Length: 0
  237.  
  238.  
  239. ---
  240. [Nov 17 07:01:47] Really destroying SIP dialog '[email protected]' Method: OPTIONS
  241. p2298758*CLI> ^V
  242. -bash-3.2#
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