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  1. 558600051@default-d532,2", "8600051|K") in new stack
  2. [Oct 23 23:53:36] WARNING[6927]: app_meetme.c:3134 admin_exec: Conference number '8600051' not found!
  3. [Oct 23 23:53:36] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-d532,2", "") in new stack
  4. [Oct 23 23:53:36] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-d532,2'
  5. [Oct 23 23:53:36] -- Executing [h@default:1] DeadAGI("Local/55558600051@default-d532,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
  6. [Oct 23 23:53:36] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  7. [Oct 23 23:53:38] == Manager 'sendcron' logged off from 127.0.0.1
  8. [Oct 23 23:53:38] == Manager 'sendcron' logged off from 127.0.0.1
  9. [Oct 23 23:53:42] Really destroying SIP dialog '[email protected]' Method: REGISTER
  10. [Oct 23 23:54:01] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:01] Found
  11. [Oct 23 23:54:01] == Manager 'sendcron' logged on from 127.0.0.1
  12. [Oct 23 23:54:01] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:01] Found
  13. [Oct 23 23:54:01] == Manager 'sendcron' logged on from 127.0.0.1
  14. [Oct 23 23:54:01] == Manager 'sendcron' logged off from 127.0.0.1
  15. [Oct 23 23:54:01] == Manager 'sendcron' logged off from 127.0.0.1
  16. [Oct 23 23:54:02] -- Executing [12503802844@default:1] AGI("SIP/gs102-0000009b", "agi://127.0.0.1:4577/call_log") in new stack
  17. [Oct 23 23:54:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
  18. [Oct 23 23:54:02] -- Executing [12503802844@default:2] Dial("SIP/gs102-0000009b", "SIP/switch2voip/12503802844|10000|To") in new stack
  19. [Oct 23 23:54:02] Audio is at 192.168.1.135 port 10368
  20. [Oct 23 23:54:02] Adding codec 0x4 (ulaw) to SDP
  21. [Oct 23 23:54:02] Adding non-codec 0x1 (telephone-event) to SDP
  22. [Oct 23 23:54:02] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  23. INVITE sip:[email protected];cpd=on SIP/2.0
  24. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;rport
  25. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  26. To: <sip:[email protected];cpd=on>
  27. Contact: <sip:[email protected]>
  28. Call-ID: [email protected]
  29. CSeq: 102 INVITE
  30. User-Agent: Asterisk PBX
  31. Max-Forwards: 70
  32. Remote-Party-ID: "Test Admin Phone" <sip:[email protected]>;privacy=off;screen=no
  33. Date: Mon, 24 Oct 2011 06:54:02 GMT
  34. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  35. Supported: replaces
  36. Content-Type: application/sdp
  37. Content-Length: 213
  38.  
  39. v=0
  40. o=root 3322 3322 IN IP4 192.168.1.135
  41. s=session
  42. c=IN IP4 192.168.1.135
  43. t=0 0
  44. m=audio 10368 RTP/AVP 0 101
  45. a=rtpmap:0 PCMU/8000
  46. a=rtpmap:101 telephone-event/8000
  47. a=fmtp:101 0-16
  48. a=ptime:20
  49. a=sendrecv
  50.  
  51. ---
  52. [Oct 23 23:54:02] -- Called switch2voip/12503802844
  53. [Oct 23 23:54:02]
  54. <--- SIP read from 66.33.147.150:5060 --->
  55. SIP/2.0 407 Unauthorized
  56. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;received=24.69.86.249;rport=5060
  57. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  58. To: <sip:[email protected];cpd=on>
  59. Call-ID: [email protected]
  60. CSeq: 102 INVITE
  61. Contact: <sip:66.33.147.150:5060>
  62. Server: Net2Phone Carrier
  63. Proxy-Authenticate: Digest realm="net2phone",nonce="B559E9EFA3C530BAB4FFC886A27A1556"
  64. Content-Length: 0
  65.  
  66.  
  67. <------------->
  68. [Oct 23 23:54:02] --- (10 headers 0 lines) ---
  69. [Oct 23 23:54:02] Transmitting (NAT) to 66.33.147.150:5060:
  70. ACK sip:[email protected];cpd=on SIP/2.0
  71. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1785cdda;rport
  72. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  73. To: <sip:[email protected];cpd=on>
  74. Contact: <sip:[email protected]>
  75. Call-ID: [email protected]
  76. CSeq: 102 ACK
  77. User-Agent: Asterisk PBX
  78. Max-Forwards: 70
  79. Remote-Party-ID: "Test Admin Phone" <sip:[email protected]>;privacy=off;screen=no
  80. Content-Length: 0
  81.  
  82.  
  83. ---
  84. [Oct 23 23:54:02] Audio is at 192.168.1.135 port 10368
  85. [Oct 23 23:54:02] Adding codec 0x4 (ulaw) to SDP
  86. [Oct 23 23:54:02] Adding non-codec 0x1 (telephone-event) to SDP
  87. [Oct 23 23:54:02] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  88. INVITE sip:[email protected];cpd=on SIP/2.0
  89. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;rport
  90. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  91. To: <sip:[email protected];cpd=on>
  92. Contact: <sip:[email protected]>
  93. Call-ID: [email protected]
  94. CSeq: 103 INVITE
  95. User-Agent: Asterisk PBX
  96. Max-Forwards: 70
  97. Remote-Party-ID: "Test Admin Phone" <sip:[email protected]>;privacy=off;screen=no
  98. Proxy-Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:[email protected];cpd=on", nonce="B559E9EFA3C530BAB4FFC886A27A1556", response="2041179d24a32fa1e36943232c172e77"
  99. Date: Mon, 24 Oct 2011 06:54:02 GMT
  100. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  101. Supported: replaces
  102. Content-Type: application/sdp
  103. Content-Length: 213
  104.  
  105. v=0
  106. o=root 3322 3323 IN IP4 192.168.1.135
  107. s=session
  108. c=IN IP4 192.168.1.135
  109. t=0 0
  110. m=audio 10368 RTP/AVP 0 101
  111. a=rtpmap:0 PCMU/8000
  112. a=rtpmap:101 telephone-event/8000
  113. a=fmtp:101 0-16
  114. a=ptime:20
  115. a=sendrecv
  116.  
  117. ---
  118. [Oct 23 23:54:02]
  119. <--- SIP read from 66.33.147.150:5060 --->
  120. SIP/2.0 100 Trying
  121. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060
  122. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  123. To: <sip:[email protected];cpd=on>;tag=ccid-188300447-1-1851
  124. Call-ID: [email protected]
  125. CSeq: 103 INVITE
  126. Contact: <sip:66.33.147.150:5060>
  127. Server: Net2Phone Carrier
  128. Content-Length: 0
  129.  
  130.  
  131. <------------->
  132. [Oct 23 23:54:02] --- (9 headers 0 lines) ---
  133. [Oct 23 23:54:04]
  134. <--- SIP read from 66.33.147.150:5060 --->
  135. SIP/2.0 183 Session Progress
  136. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060
  137. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  138. To: <sip:[email protected];cpd=on>;tag=ccid-188300447-1-1851
  139. Call-ID: [email protected]
  140. CSeq: 103 INVITE
  141. Contact: <sip:66.33.147.150:5060>
  142. Server: Net2Phone Carrier
  143. Content-Length: 214
  144. Content-Type: application/sdp
  145.  
  146. v=0
  147. o=5208913412 188300447 188300447 IN IP4 216.53.10.3
  148. s=SIP Call
  149. c=IN IP4 216.53.10.3
  150. t=0 0
  151. m=audio 20798 RTP/AVP 0 101
  152. a=ptime:20
  153. a=rtpmap:0 PCMU/8000
  154. a=rtpmap:101 telephone-event/8000
  155. a=fmtp:101 0-11
  156.  
  157. <------------->
  158. [Oct 23 23:54:04] --- (10 headers 10 lines) ---
  159. [Oct 23 23:54:04] Found RTP audio format 0
  160. [Oct 23 23:54:04] Found RTP audio format 101
  161. [Oct 23 23:54:04] Found audio description format PCMU for ID 0
  162. [Oct 23 23:54:04] Found audio description format telephone-event for ID 101
  163. [Oct 23 23:54:04] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  164. [Oct 23 23:54:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  165. [Oct 23 23:54:04] Peer audio RTP is at port 216.53.10.3:20798
  166. [Oct 23 23:54:04] -- SIP/switch2voip-0000009c is making progress passing it to SIP/gs102-0000009b
  167. [Oct 23 23:54:05]
  168. <--- SIP read from 66.33.147.150:5060 --->
  169. SIP/2.0 200 OK
  170. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4eb5dec5;received=24.69.86.249;rport=5060
  171. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  172. To: <sip:[email protected];cpd=on>;tag=ccid-188300447-1-1851
  173. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS
  174. Call-ID: [email protected]
  175. CSeq: 103 INVITE
  176. Contact: <sip:66.33.147.150:5060>
  177. Server: Net2Phone Carrier
  178. Content-Length: 214
  179. ontent-Type: application/sdp
  180.  
  181. v=0
  182. o=5208913412 188300447 188300447 IN IP4 216.53.10.3
  183. s=SIP Call
  184. c=IN IP4 216.53.10.3
  185. t=0 0
  186. m=audio 20798 RTP/AVP 0 101
  187. a=ptime:20
  188. a=rtpmap:0 PCMU/8000
  189. a=rtpmap:101 telephone-event/8000
  190. a=fmtp:101 0-11
  191.  
  192. <------------->
  193. [Oct 23 23:54:05] --- (11 headers 10 lines) ---
  194. [Oct 23 23:54:05] Found RTP audio format 0
  195. [Oct 23 23:54:05] Found RTP audio format 101
  196. [Oct 23 23:54:05] Found audio description format PCMU for ID 0
  197. [Oct 23 23:54:05] Found audio description format telephone-event for ID 101
  198. [Oct 23 23:54:05] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  199. [Oct 23 23:54:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  200. [Oct 23 23:54:05] Peer audio RTP is at port 216.53.10.3:20798
  201. [Oct 23 23:54:05] list_route: hop: <sip:66.33.147.150:5060>
  202. [Oct 23 23:54:05] set_destination: Parsing <sip:66.33.147.150:5060> for address/port to send to
  203. [Oct 23 23:54:05] set_destination: set destination to 66.33.147.150, port 5060
  204. [Oct 23 23:54:05] Transmitting (NAT) to 66.33.147.150:5060:
  205. ACK sip:66.33.147.150:5060 SIP/2.0
  206. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK2400bf41;rport
  207. From: "Test Admin Phone" <sip:[email protected]>;tag=as4f7f71f2
  208. To: <sip:[email protected];cpd=on>;tag=ccid-188300447-1-1851
  209. Contact: <sip:[email protected]>
  210. Call-ID: [email protected]
  211. CSeq: 103 ACK
  212. User-Agent: Asterisk PBX
  213. Max-Forwards: 70
  214. Remote-Party-ID: "Test Admin Phone" <sip:[email protected]>;privacy=off;screen=no
  215. Content-Length: 0
  216.  
  217.  
  218. ---
  219. [Oct 23 23:54:05] -- SIP/switch2voip-0000009c answered SIP/gs102-0000009b
  220. [Oct 23 23:54:06] == Parsing '/etc/asterisk/manager.conf': [Oct 23 23:54:06] Found
  221. [Oct 23 23:54:06] == Manager 'sendcron' logged on from 127.0.0.1
  222. [Oct 23 23:54:06] == Manager 'sendcron' logged off from 127.0.0.1
  223. [Oct 23 23:54:10] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  224. OPTIONS sip:66.33.147.150;cpd=on SIP/2.0
  225. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1e9cab9b;rport
  226. From: "asterisk" <sip:[email protected]>;tag=as50188c95
  227. To: <sip:66.33.147.150;cpd=on>
  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: Mon, 24 Oct 2011 06:54:10 GMT
  234. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  235. Supported: replaces
  236. Content-Length: 0
  237.  
  238.  
  239. ---
  240. [Oct 23 23:54:10]
  241. <--- SIP read from 66.33.147.150:5060 --->
  242. SIP/2.0 200 OK
  243. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK1e9cab9b;received=24.69.86.249;rport=5060
  244. From: "asterisk" <sip:[email protected]>;tag=as50188c95
  245. To: <sip:66.33.147.150;cpd=on>
  246. Call-ID: [email protected]
  247. CSeq: 102 OPTIONS
  248. Contact: <sip:66.33.147.150:5060>
  249. Server: Net2Phone Carrier
  250. Content-Length: 0
  251.  
  252.  
  253. <------------->
  254. [Oct 23 23:54:10] --- (9 headers 0 lines) ---
  255. [Oct 23 23:54:10] Really destroying SIP dialog '[email protected]' Method: OPTIONS
  256. vicibox*CLI>
  257.  
  258.  
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