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  1. [Jan 26 08:41:53] VERBOSE[23749] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 26 08:41:53] DEBUG[23749] config.c: Parsing /etc/asterisk/logger.conf
  2. [Jan 26 08:41:53] VERBOSE[23749] config.c: == Found
  3. [Jan 26 08:41:53] VERBOSE[23749] logger.c: Asterisk Queue Logger restarted
  4. [Jan 26 08:42:15] VERBOSE[23737] chan_sip.c:
  5. <--- SIP read from UDP:212.7.117.61:48052 --->
  6.  
  7.  
  8. <------------->
  9. [Jan 26 08:42:15] DEBUG[23737] chan_sip.c: Header 0 [ 0]:
  10. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  11. <--- SIP read from UDP:212.7.117.61:48052 --->
  12. INVITE sip:10000009@mypbx.mydomain.com SIP/2.0
  13. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  14. Max-Forwards: 70
  15. Contact: <sip:dovid@212.7.117.61:48052>
  16. To: "10000009"<sip:10000009@mypbx.mydomain.com>
  17. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  18. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  19. CSeq: 1 INVITE
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  21. Content-Type: application/sdp
  22. User-Agent: eyeBeam release 1102q stamp 51814
  23. Content-Length: 334
  24.  
  25. v=0
  26. o=- 9 2 IN IP4 192.168.1.10
  27. s=CounterPath eyeBeam 1.5
  28. c=IN IP4 192.168.1.10
  29. t=0 0
  30. m=audio 53352 RTP/AVP 9 0 18 101
  31. a=alt:1 2 : uf2tRT5+ wgYeAWAo 192.168.1.10 53352
  32. a=alt:2 1 : QPFLPL8S 6JvRvgDl 192.168.56.1 53352
  33. a=fmtp:18 annexb=yes
  34. a=fmtp:101 0-15
  35. a=rtpmap:18 G729/8000
  36. a=rtpmap:101 telephone-event/8000
  37. a=sendrecv
  38. <------------->
  39. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 48]: INVITE sip:10000009@mypbx.mydomain.com SIP/2.0
  40. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  41. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  42. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  43. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 49]: To: "10000009"<sip:10000009@mypbx.mydomain.com>
  44. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  45. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  46. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 7 [ 14]: CSeq: 1 INVITE
  47. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  48. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp
  49. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 10 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  50. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 11 [ 19]: Content-Length: 334
  51. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 12 [ 0]:
  52. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  53. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 9 2 IN IP4 192.168.1.10
  54. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5
  55. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.10
  56. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  57. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 5 [ 32]: m=audio 53352 RTP/AVP 9 0 18 101
  58. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 6 [ 48]: a=alt:1 2 : uf2tRT5+ wgYeAWAo 192.168.1.10 53352
  59. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 7 [ 48]: a=alt:2 1 : QPFLPL8S 6JvRvgDl 192.168.56.1 53352
  60. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 8 [ 20]: a=fmtp:18 annexb=yes
  61. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15
  62. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
  63. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000
  64. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 12 [ 10]: a=sendrecv
  65. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (12 headers 13 lines) ---
  66. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag
  67. [Jan 26 08:42:22] DEBUG[23737] acl.c: For destination '212.7.117.61', our source address is '208.211.92.75'.
  68. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 208.211.92.75:5060
  69. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Allocating new SIP dialog for Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. - INVITE (No RTP)
  70. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  71. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:48052' gives...
  72. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '48052'.
  73. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:48052 (no NAT)
  74. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Initializing initreq for method INVITE - callid Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  75. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Using INVITE request as basis request - Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  76. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found peer 'dovid' for 'dovid' from 212.7.117.61:48052
  77. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  78. <--- Reliably Transmitting (NAT) to 212.7.117.61:48052 --->
  79. SIP/2.0 401 Unauthorized
  80. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;received=212.7.117.61;rport=48052
  81. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  82. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as3df137b7
  83. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  84. CSeq: 1 INVITE
  85. Server: Asterisk PBX 1.8.2.2
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  87. Supported: replaces, timer
  88. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1da99604"
  89. Content-Length: 0
  90.  
  91.  
  92. <------------>
  93. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #102
  94. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.7.117.61:48052
  95. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' in 32000 ms (Method: INVITE)
  96. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  97. <--- SIP read from UDP:212.7.117.61:48052 --->
  98. ACK sip:10000009@mypbx.mydomain.com SIP/2.0
  99. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  100. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as3df137b7
  101. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  102. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  103. CSeq: 1 ACK
  104. Content-Length: 0
  105.  
  106. <------------->
  107. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 45]: ACK sip:10000009@mypbx.mydomain.com SIP/2.0
  108. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  109. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 64]: To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as3df137b7
  110. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  111. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  112. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK
  113. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 17]: Content-Length: 0
  114. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (7 headers 0 lines) ---
  115. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag as3df137b7
  116. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  117. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #102
  118. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Stopping retransmission on 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' of Response 1: Match Found
  119. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  120. <--- SIP read from UDP:212.7.117.61:48052 --->
  121. ACK sip:10000009@mypbx.mydomain.com SIP/2.0
  122. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  123. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as3df137b7
  124. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  125. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  126. CSeq: 1 ACK
  127. Content-Length: 0
  128.  
  129. <------------->
  130. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 45]: ACK sip:10000009@mypbx.mydomain.com SIP/2.0
  131. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-cd083f5f0223017a-1---d8754z-;rport
  132. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 64]: To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as3df137b7
  133. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  134. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  135. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK
  136. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 17]: Content-Length: 0
  137. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (7 headers 0 lines) ---
  138. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag as3df137b7
  139. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  140. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  141. <--- SIP read from UDP:212.7.117.61:48052 --->
  142. INVITE sip:10000009@mypbx.mydomain.com SIP/2.0
  143. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-7b160400804d463f-1---d8754z-;rport
  144. Max-Forwards: 70
  145. Contact: <sip:dovid@212.7.117.61:48052>
  146. To: "10000009"<sip:10000009@mypbx.mydomain.com>
  147. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  148. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  149. CSeq: 2 INVITE
  150. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  151. Content-Type: application/sdp
  152. User-Agent: eyeBeam release 1102q stamp 51814
  153. Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  154. Content-Length: 334
  155.  
  156. v=0
  157. o=- 9 2 IN IP4 192.168.1.10
  158. s=CounterPath eyeBeam 1.5
  159. c=IN IP4 192.168.1.10
  160. t=0 0
  161. m=audio 53352 RTP/AVP 9 0 18 101
  162. a=alt:1 2 : uf2tRT5+ wgYeAWAo 192.168.1.10 53352
  163. a=alt:2 1 : QPFLPL8S 6JvRvgDl 192.168.56.1 53352
  164. a=fmtp:18 annexb=yes
  165. a=fmtp:101 0-15
  166. a=rtpmap:18 G729/8000
  167. a=rtpmap:101 telephone-event/8000
  168. a=sendrecv
  169. <------------->
  170. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 48]: INVITE sip:10000009@mypbx.mydomain.com SIP/2.0
  171. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-7b160400804d463f-1---d8754z-;rport
  172. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  173. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  174. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 49]: To: "10000009"<sip:10000009@mypbx.mydomain.com>
  175. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  176. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  177. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE
  178. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  179. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp
  180. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 10 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  181. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 11 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  182. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 334
  183. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 13 [ 0]:
  184. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  185. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 9 2 IN IP4 192.168.1.10
  186. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5
  187. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.10
  188. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  189. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 5 [ 32]: m=audio 53352 RTP/AVP 9 0 18 101
  190. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 6 [ 48]: a=alt:1 2 : uf2tRT5+ wgYeAWAo 192.168.1.10 53352
  191. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 7 [ 48]: a=alt:2 1 : QPFLPL8S 6JvRvgDl 192.168.56.1 53352
  192. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 8 [ 20]: a=fmtp:18 annexb=yes
  193. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15
  194. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
  195. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000
  196. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Body 12 [ 10]: a=sendrecv
  197. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) ---
  198. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag
  199. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: Splitting 'mypbx.mydomain.com' gives...
  200. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: ...host 'mypbx.mydomain.com' and port '(null)'.
  201. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: Splitting 'mypbx.mydomain.com' gives...
  202. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: ...host 'mypbx.mydomain.com' and port '(null)'.
  203. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  204. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:48052' gives...
  205. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '48052'.
  206. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:48052 (NAT)
  207. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Initializing initreq for method INVITE - callid Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  208. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Using INVITE request as basis request - Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  209. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found peer 'dovid' for 'dovid' from 212.7.117.61:48052
  210. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd050c00'
  211. [Jan 26 08:42:22] DEBUG[23737] res_rtp_asterisk.c: Allocated port 19710 for RTP instance '0xd050c00'
  212. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: RTP instance '0xd050c00' is setup and ready to go
  213. [Jan 26 08:42:22] DEBUG[23737] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd050c00'
  214. [Jan 26 08:42:22] VERBOSE[23737] netsock2.c: == Using SIP RTP TOS bits 184
  215. [Jan 26 08:42:22] VERBOSE[23737] netsock2.c: == Using SIP RTP CoS mark 5
  216. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Setting NAT on RTP to On
  217. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  218. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing session-level SDP o=- 9 2 IN IP4 192.168.1.10... UNSUPPORTED.
  219. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED.
  220. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: Splitting '192.168.1.10' gives...
  221. [Jan 26 08:42:22] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '(null)'.
  222. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.10... OK.
  223. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  224. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found RTP audio format 9
  225. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Setting payload 9 based on m type on 0xb4508490
  226. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  227. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508490
  228. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found RTP audio format 18
  229. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Setting payload 18 based on m type on 0xb4508490
  230. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  231. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508490
  232. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=alt:1 2 : uf2tRT5+ wgYeAWAo 192.168.1.10 53352... UNSUPPORTED.
  233. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=alt:2 1 : QPFLPL8S 6JvRvgDl 192.168.56.1 53352... UNSUPPORTED.
  234. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED.
  235. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
  236. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found audio description format G729 for ID 18
  237. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
  238. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  239. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  240. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  241. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508490
  242. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Incorporating payload 9 on 0xb4508490
  243. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Incorporating payload 18 on 0xb4508490
  244. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508490
  245. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1104 (ulaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  246. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  247. [Jan 26 08:42:22] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  248. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 192.168.1.10:53352
  249. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508490 to 0xd050dac
  250. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Copying payload 9 from 0xb4508490 to 0xd050dac
  251. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Copying payload 18 from 0xb4508490 to 0xd050dac
  252. [Jan 26 08:42:22] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508490 to 0xd050dac
  253. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
  254. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Checking SIP call limits for device dovid
  255. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Updating call counter for incoming call
  256. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Looking for 10000009 in dovid (domain mypbx.mydomain.com)
  257. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  258. Event: Newchannel
  259. Privilege: call,all
  260. Channel: SIP/dovid-0000000a
  261. ChannelState: 0
  262. ChannelStateDesc: Down
  263. CallerIDNum: dovid
  264. CallerIDName: dovid
  265. AccountCode:
  266. Exten: 10000009
  267. Context: dovid
  268. Uniqueid: 1296049342.10
  269.  
  270.  
  271. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** Our native formats are 0x4 (ulaw)
  272. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** Joint capabilities are 0x4 (ulaw)
  273. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw)
  274. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
  275. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: This channel will not be able to handle video.
  276. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: build_route: Contact hop: <sip:dovid@212.7.117.61:48052>
  277. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: list_route: hop: <sip:dovid@212.7.117.61:48052>
  278. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: SIP/dovid-0000000a: New call is still down.... Trying...
  279. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  280. <--- Transmitting (NAT) to 212.7.117.61:48052 --->
  281. SIP/2.0 100 Trying
  282. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-7b160400804d463f-1---d8754z-;received=212.7.117.61;rport=48052
  283. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  284. To: "10000009"<sip:10000009@mypbx.mydomain.com>
  285. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  286. CSeq: 2 INVITE
  287. Server: Asterisk PBX 1.8.2.2
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  289. Supported: replaces, timer
  290. Contact: <sip:10000009@208.211.92.75:5060>
  291. Content-Length: 0
  292.  
  293.  
  294. <------------>
  295. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 212.7.117.61:48052
  296. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  297. Event: VarSet
  298. Privilege: dialplan,all
  299. Channel: SIP/dovid-0000000a
  300. Variable: SIPURI
  301. Value: sip:dovid@212.7.117.61:48052
  302. Uniqueid: 1296049342.10
  303.  
  304.  
  305. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  306. Event: VarSet
  307. Privilege: dialplan,all
  308. Channel: SIP/dovid-0000000a
  309. Variable: SIPDOMAIN
  310. Value: mypbx.mydomain.com
  311. Uniqueid: 1296049342.10
  312.  
  313.  
  314. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  315. Event: VarSet
  316. Privilege: dialplan,all
  317. Channel: SIP/dovid-0000000a
  318. Variable: SIPCALLID
  319. Value: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  320. Uniqueid: 1296049342.10
  321.  
  322.  
  323. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  324. Event: Newstate
  325. Privilege: call,all
  326. Channel: SIP/dovid-0000000a
  327. ChannelState: 4
  328. ChannelStateDesc: Ring
  329. CallerIDNum: dovid
  330. CallerIDName: dovid
  331. Uniqueid: 1296049342.10
  332.  
  333.  
  334. [Jan 26 08:42:22] DEBUG[23821] pbx.c: Result of 'EXTEN' is '10000009'
  335. [Jan 26 08:42:22] DEBUG[23821] pbx.c: Launching 'Dial'
  336. [Jan 26 08:42:22] VERBOSE[23821] pbx.c: -- Executing [10000009@dovid:1] Dial("SIP/dovid-0000000a", "SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)") in new stack
  337. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  338. Event: Newexten
  339. Privilege: dialplan,all
  340. Channel: SIP/dovid-0000000a
  341. Context: dovid
  342. Extension: 10000009
  343. Priority: 1
  344. Application: Dial
  345. AppData: SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)
  346. Uniqueid: 1296049342.10
  347.  
  348.  
  349. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  350. Event: VarSet
  351. Privilege: dialplan,all
  352. Channel: SIP/dovid-0000000a
  353. Variable: DIALSTATUS
  354. Value:
  355. Uniqueid: 1296049342.10
  356.  
  357.  
  358. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  359. Event: VarSet
  360. Privilege: dialplan,all
  361. Channel: SIP/dovid-0000000a
  362. Variable: DIALEDPEERNUMBER
  363. Value:
  364. Uniqueid: 1296049342.10
  365.  
  366.  
  367. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  368. Event: VarSet
  369. Privilege: dialplan,all
  370. Channel: SIP/dovid-0000000a
  371. Variable: DIALEDPEERNAME
  372. Value:
  373. Uniqueid: 1296049342.10
  374.  
  375.  
  376. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  377. Event: VarSet
  378. Privilege: dialplan,all
  379. Channel: SIP/dovid-0000000a
  380. Variable: ANSWEREDTIME
  381. Value:
  382. Uniqueid: 1296049342.10
  383.  
  384.  
  385. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  386. Event: VarSet
  387. Privilege: dialplan,all
  388. Channel: SIP/dovid-0000000a
  389. Variable: DIALEDTIME
  390. Value:
  391. Uniqueid: 1296049342.10
  392.  
  393.  
  394. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
  395. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Allocating new SIP dialog for 0f00afe86433f58212d27b2d1277ad55@208.211.92.75:0 - INVITE (No RTP)
  396. [Jan 26 08:42:22] DEBUG[23821] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd055c98'
  397. [Jan 26 08:42:22] DEBUG[23821] res_rtp_asterisk.c: Allocated port 14076 for RTP instance '0xd055c98'
  398. [Jan 26 08:42:22] DEBUG[23821] rtp_engine.c: RTP instance '0xd055c98' is setup and ready to go
  399. [Jan 26 08:42:22] DEBUG[23821] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd055c98'
  400. [Jan 26 08:42:22] VERBOSE[23821] netsock2.c: == Using SIP RTP TOS bits 184
  401. [Jan 26 08:42:22] VERBOSE[23821] netsock2.c: == Using SIP RTP CoS mark 5
  402. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Setting NAT on RTP to Off
  403. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: OBPROXY: Not applying OBproxy to this call
  404. [Jan 26 08:42:22] DEBUG[23821] acl.c: For destination '69.167.68.130', our source address is '208.211.92.75'.
  405. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 208.211.92.75:5060
  406. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  407. Event: Newchannel
  408. Privilege: call,all
  409. Channel: SIP/fpp-0000000b
  410. ChannelState: 0
  411. ChannelStateDesc: Down
  412. CallerIDNum:
  413. CallerIDName:
  414. AccountCode:
  415. Exten:
  416. Context: from-sip
  417. Uniqueid: 1296049342.11
  418.  
  419.  
  420. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** Our native formats are 0x4 (ulaw)
  421. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** Joint capabilities are 0x4 (ulaw)
  422. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw)
  423. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
  424. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
  425. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: This channel will not be able to handle video.
  426. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  427. Event: VarSet
  428. Privilege: dialplan,all
  429. Channel: SIP/fpp-0000000b
  430. Variable: SIPCALLID
  431. Value: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  432. Uniqueid: 1296049342.11
  433.  
  434.  
  435. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  436. Event: VarSet
  437. Privilege: dialplan,all
  438. Channel: SIP/fpp-0000000b
  439. Variable: DIALEDPEERNUMBER
  440. Value: 10000009@fpp
  441. Uniqueid: 1296049342.11
  442.  
  443.  
  444. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable DIALEDTIME.
  445. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable ANSWEREDTIME.
  446. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable DIALEDPEERNAME.
  447. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable DIALEDPEERNUMBER.
  448. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable DIALSTATUS.
  449. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable SIPCALLID.
  450. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable SIPDOMAIN.
  451. [Jan 26 08:42:22] DEBUG[23821] channel.c: Not copying variable SIPURI.
  452. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Outgoing Call for 10000009
  453. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Updating call counter for outgoing call
  454. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False
  455. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  456. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c: Audio is at 5060
  457. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  458. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  459. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  460. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  461. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
  462. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Initializing initreq for method INVITE - callid 730bfb20211d6c7a40e584041062e145@69.167.68.130
  463. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 0 [ 41]: INVITE sip:10000009@69.167.68.130 SIP/2.0
  464. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK06e3dc41
  465. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  466. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  467. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 4 [ 32]: To: <sip:10000009@69.167.68.130>
  468. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 5 [ 39]: Contact: <sip:dovid@208.211.92.75:5060>
  469. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 6 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  470. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
  471. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  472. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 9 [ 35]: Date: Wed, 26 Jan 2011 13:42:22 GMT
  473. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  474. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
  475. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
  476. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  477. INVITE sip:10000009@69.167.68.130 SIP/2.0
  478. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK06e3dc41
  479. Max-Forwards: 70
  480. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  481. To: <sip:10000009@69.167.68.130>
  482. Contact: <sip:dovid@208.211.92.75:5060>
  483. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  484. CSeq: 102 INVITE
  485. User-Agent: Asterisk PBX 1.8.2.2
  486. Date: Wed, 26 Jan 2011 13:42:22 GMT
  487. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  488. Supported: replaces, timer
  489. Content-Type: application/sdp
  490. Content-Length: 289
  491.  
  492. v=0
  493. o=root 1174122120 1174122120 IN IP4 208.211.92.75
  494. s=Asterisk PBX 1.8.2.2
  495. c=IN IP4 208.211.92.75
  496. t=0 0
  497. m=audio 14076 RTP/AVP 0 8 101
  498. a=rtpmap:0 PCMU/8000
  499. a=rtpmap:8 PCMA/8000
  500. a=rtpmap:101 telephone-event/8000
  501. a=fmtp:101 0-16
  502. a=silenceSupp:off - - - -
  503. a=ptime:20
  504. a=sendrecv
  505.  
  506. ---
  507. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #105
  508. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  509. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  510. Event: Dial
  511. Privilege: call,all
  512. SubEvent: Begin
  513. Channel: SIP/dovid-0000000a
  514. Destination: SIP/fpp-0000000b
  515. CallerIDNum: dovid
  516. CallerIDName: dovid
  517. UniqueID: 1296049342.10
  518. DestUniqueID: 1296049342.11
  519. Dialstring: 10000009@fpp
  520.  
  521.  
  522. [Jan 26 08:42:22] VERBOSE[23821] app_dial.c: -- Called 10000009@fpp
  523. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  524. Event: NewCallerid
  525. Privilege: call,all
  526. Channel: SIP/fpp-0000000b
  527. CallerIDNum: 10000009
  528. CallerIDName:
  529. Uniqueid: 1296049342.11
  530. CID-CallingPres: 0 (Presentation Allowed, Not Screened)
  531.  
  532.  
  533. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid
  534. [Jan 26 08:42:22] DEBUG[23716] chan_sip.c: Checking device state for peer dovid
  535. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use)
  536. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1'
  537. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  538. <--- SIP read from UDP:69.167.68.130:5060 --->
  539. SIP/2.0 407 Proxy Authentication Required
  540. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK06e3dc41
  541. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  542. To: <sip:10000009@69.167.68.130>;tag=8a7940c898c7113a1fb8a4a76e6676f0.ba1f
  543. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  544. CSeq: 102 INVITE
  545. Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024dc00009ad68a3ae8028c24e55d82e050d08dc7b946"
  546. Server: PBX_MANAGER
  547. Content-Length: 0
  548. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.130 via_cnt==1"
  549.  
  550. <------------->
  551. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required
  552. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK06e3dc41
  553. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  554. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 74]: To: <sip:10000009@69.167.68.130>;tag=8a7940c898c7113a1fb8a4a76e6676f0.ba1f
  555. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  556. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
  557. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024dc00009ad68a3ae8028c24e55d82e050d08dc7b946"
  558. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  559. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0
  560. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 9 [188]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.130 via_cnt==1"
  561. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) ---
  562. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag 8a7940c898c7113a1fb8a4a76e6676f0.ba1f
  563. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Acked pending invite 102
  564. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #105
  565. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 102: Match Found
  566. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: SIP response 407 to standard invite
  567. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  568. ACK sip:10000009@69.167.68.130 SIP/2.0
  569. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK06e3dc41
  570. Max-Forwards: 70
  571. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  572. To: <sip:10000009@69.167.68.130>;tag=8a7940c898c7113a1fb8a4a76e6676f0.ba1f
  573. Contact: <sip:dovid@208.211.92.75:5060>
  574. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  575. CSeq: 102 ACK
  576. User-Agent: Asterisk PBX 1.8.2.2
  577. Content-Length: 0
  578.  
  579.  
  580. ---
  581. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  582. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE
  583. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False
  584. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  585. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Audio is at 5060
  586. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  587. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  588. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  589. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP
  590. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
  591. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  592. INVITE sip:10000009@69.167.68.130 SIP/2.0
  593. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  594. Max-Forwards: 70
  595. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  596. To: <sip:10000009@69.167.68.130>
  597. Contact: <sip:dovid@208.211.92.75:5060>
  598. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  599. CSeq: 103 INVITE
  600. User-Agent: Asterisk PBX 1.8.2.2
  601. Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.130", nonce="4d4024dc00009ad68a3ae8028c24e55d82e050d08dc7b946", response="0ff18ca1556bb75fe94a1229e021fc1d"
  602. Date: Wed, 26 Jan 2011 13:42:22 GMT
  603. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  604. Supported: replaces, timer
  605. Content-Type: application/sdp
  606. Content-Length: 289
  607.  
  608. v=0
  609. o=root 1174122120 1174122121 IN IP4 208.211.92.75
  610. s=Asterisk PBX 1.8.2.2
  611. c=IN IP4 208.211.92.75
  612. t=0 0
  613. m=audio 14076 RTP/AVP 0 8 101
  614. a=rtpmap:0 PCMU/8000
  615. a=rtpmap:8 PCMA/8000
  616. a=rtpmap:101 telephone-event/8000
  617. a=fmtp:101 0-16
  618. a=silenceSupp:off - - - -
  619. a=ptime:20
  620. a=sendrecv
  621.  
  622. ---
  623. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #107
  624. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  625. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  626. Event: VarSet
  627. Privilege: dialplan,all
  628. Channel: SIP/dovid-0000000a
  629. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  630. Value: SIP 407 Proxy Authentication Required
  631. Uniqueid: 1296049342.10
  632.  
  633.  
  634. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  635. <--- SIP read from UDP:69.167.68.130:5060 --->
  636. SIP/2.0 100 Giving a try
  637. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  638. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  639. To: <sip:10000009@69.167.68.130>
  640. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  641. CSeq: 103 INVITE
  642. Server: PBX_MANAGER
  643. Content-Length: 0
  644. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.133:5060;transport=udp via_cnt==1"
  645.  
  646. <------------->
  647. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try
  648. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  649. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  650. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 32]: To: <sip:10000009@69.167.68.130>
  651. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  652. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE
  653. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER
  654. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  655. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 8 [207]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.133:5060;transport=udp via_cnt==1"
  656. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) ---
  657. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag
  658. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #107 - INVITE (got response)
  659. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '730bfb20211d6c7a40e584041062e145@69.167.68.130' Request 103: Found
  660. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: SIP response 100 to standard invite
  661. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  662. Event: VarSet
  663. Privilege: dialplan,all
  664. Channel: SIP/dovid-0000000a
  665. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  666. Value: SIP 100 Giving a try
  667. Uniqueid: 1296049342.10
  668.  
  669.  
  670. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c:
  671. <--- SIP read from UDP:69.167.68.130:5060 --->
  672. SIP/2.0 180 Ringing
  673. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  674. Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  675. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  676. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  677. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  678. CSeq: 103 INVITE
  679. Server: PBX_MANAGER
  680. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  681. Supported: replaces, timer
  682. Contact: <sip:10000009@69.167.68.133:5060>
  683. Content-Length: 0
  684.  
  685. <------------->
  686. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing
  687. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  688. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  689. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  690. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  691. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  692. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE
  693. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  694. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  695. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer
  696. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: <sip:10000009@69.167.68.133:5060>
  697. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: Header 11 [ 17]: Content-Length: 0
  698. [Jan 26 08:42:22] VERBOSE[23737] chan_sip.c: --- (12 headers 0 lines) ---
  699. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  700. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '730bfb20211d6c7a40e584041062e145@69.167.68.130' Request 103: Found
  701. [Jan 26 08:42:22] DEBUG[23737] chan_sip.c: SIP response 180 to standard invite
  702. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  703. Event: Newstate
  704. Privilege: call,all
  705. Channel: SIP/fpp-0000000b
  706. ChannelState: 5
  707. ChannelStateDesc: Ringing
  708. CallerIDNum: 10000009
  709. CallerIDName:
  710. Uniqueid: 1296049342.11
  711.  
  712.  
  713. [Jan 26 08:42:22] DEBUG[23750] manager.c: Examining event:
  714. Event: VarSet
  715. Privilege: dialplan,all
  716. Channel: SIP/dovid-0000000a
  717. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  718. Value: SIP 180 Ringing
  719. Uniqueid: 1296049342.10
  720.  
  721.  
  722. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp
  723. [Jan 26 08:42:22] DEBUG[23716] chan_sip.c: Checking device state for peer fpp
  724. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use)
  725. [Jan 26 08:42:22] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1'
  726. [Jan 26 08:42:22] VERBOSE[23821] app_dial.c: -- SIP/fpp-0000000b is ringing
  727. [Jan 26 08:42:22] VERBOSE[23821] chan_sip.c:
  728. <--- Transmitting (NAT) to 212.7.117.61:48052 --->
  729. SIP/2.0 180 Ringing
  730. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-7b160400804d463f-1---d8754z-;received=212.7.117.61;rport=48052
  731. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  732. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  733. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  734. CSeq: 2 INVITE
  735. Server: Asterisk PBX 1.8.2.2
  736. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  737. Supported: replaces, timer
  738. Contact: <sip:10000009@208.211.92.75:5060>
  739. Content-Length: 0
  740.  
  741.  
  742. <------------>
  743. [Jan 26 08:42:22] DEBUG[23821] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 212.7.117.61:48052
  744. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c:
  745. <--- SIP read from UDP:69.167.68.130:5060 --->
  746. SIP/2.0 200 OK
  747. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  748. Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  749. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  750. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  751. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  752. CSeq: 103 INVITE
  753. Server: PBX_MANAGER
  754. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  755. Supported: replaces, timer
  756. Contact: <sip:10000009@69.167.68.133:5060>
  757. Content-Type: application/sdp
  758. Content-Length: 298
  759.  
  760. v=0
  761. o=root 918636038 918636038 IN IP4 69.167.68.133
  762. s=PBX_MANAGER
  763. c=IN IP4 69.167.68.133
  764. t=0 0
  765. m=audio 15760 RTP/AVP 0 8 101
  766. a=rtpmap:0 PCMU/8000
  767. a=rtpmap:8 PCMA/8000
  768. a=rtpmap:101 telephone-event/8000
  769. a=fmtp:101 0-16
  770. a=silenceSupp:off - - - -
  771. a=ptime:20
  772. a=sendrecv
  773. a=direction:active
  774. <------------->
  775. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  776. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK57e7cbbf
  777. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  778. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  779. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  780. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  781. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE
  782. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  783. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  784. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer
  785. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: <sip:10000009@69.167.68.133:5060>
  786. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp
  787. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 298
  788. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Header 13 [ 0]:
  789. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  790. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 1 [ 47]: o=root 918636038 918636038 IN IP4 69.167.68.133
  791. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER
  792. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.133
  793. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  794. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 5 [ 29]: m=audio 15760 RTP/AVP 0 8 101
  795. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
  796. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000
  797. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000
  798. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16
  799. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 10 [ 25]: a=silenceSupp:off - - - -
  800. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=ptime:20
  801. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 12 [ 10]: a=sendrecv
  802. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Body 13 [ 18]: a=direction:active
  803. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: --- (13 headers 14 lines) ---
  804. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  805. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Acked pending invite 103
  806. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 103: Match Found
  807. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: SIP response 200 to standard invite
  808. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  809. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 918636038 918636038 IN IP4 69.167.68.133... UNSUPPORTED.
  810. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED.
  811. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: Splitting '69.167.68.133' gives...
  812. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '(null)'.
  813. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.133... OK.
  814. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  815. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  816. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100
  817. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found RTP audio format 8
  818. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Setting payload 8 based on m type on 0xb4508100
  819. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  820. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100
  821. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0
  822. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  823. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found audio description format PCMA for ID 8
  824. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
  825. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  826. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  827. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
  828. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
  829. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  830. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  831. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED.
  832. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100
  833. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Incorporating payload 8 on 0xb4508100
  834. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100
  835. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  836. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  837. [Jan 26 08:42:25] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  838. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.133:15760
  839. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd055e44
  840. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Copying payload 8 from 0xb4508100 to 0xd055e44
  841. [Jan 26 08:42:25] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd055e44
  842. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw)
  843. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call
  844. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call
  845. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: build_route: Record-Route hop: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  846. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: list_route: hop: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  847. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: Splitting '69.167.68.133:5060' gives...
  848. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '5060'.
  849. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  850. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  851. [Jan 26 08:42:25] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  852. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  853. [Jan 26 08:42:25] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  854. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  855. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6c3cd24c
  856. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  857. Max-Forwards: 70
  858. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  859. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  860. Contact: <sip:dovid@208.211.92.75:5060>
  861. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  862. CSeq: 103 ACK
  863. User-Agent: Asterisk PBX 1.8.2.2
  864. Content-Length: 0
  865.  
  866.  
  867. ---
  868. [Jan 26 08:42:25] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  869. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  870. Event: VarSet
  871. Privilege: dialplan,all
  872. Channel: SIP/dovid-0000000a
  873. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  874. Value: SIP 200 OK
  875. Uniqueid: 1296049342.10
  876.  
  877.  
  878. [Jan 26 08:42:25] VERBOSE[23821] app_dial.c: -- SIP/fpp-0000000b answered SIP/dovid-0000000a
  879. [Jan 26 08:42:25] DEBUG[23821] app_stack.c: Channel SIP/fpp-0000000b has no datastore, so we're allocating one.
  880. [Jan 26 08:42:25] DEBUG[23821] app_stack.c: Setting 'ARG1' to 's'
  881. [Jan 26 08:42:25] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp
  882. [Jan 26 08:42:25] DEBUG[23821] app_stack.c: Setting 'ARG2' to '1'
  883. [Jan 26 08:42:25] DEBUG[23716] chan_sip.c: Checking device state for peer fpp
  884. [Jan 26 08:42:25] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use)
  885. [Jan 26 08:42:25] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1'
  886. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Launching 'AGI'
  887. [Jan 26 08:42:25] VERBOSE[23821] pbx.c: -- Executing [s@do_dtmf_cc-take-call:1] AGI("SIP/fpp-0000000b", "agi://127.0.0.1:4579/update_call_status?status=60") in new stack
  888. [Jan 26 08:42:25] DEBUG[23821] res_agi.c: Wow, connected!
  889. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  890. Event: Newstate
  891. Privilege: call,all
  892. Channel: SIP/fpp-0000000b
  893. ChannelState: 6
  894. ChannelStateDesc: Up
  895. CallerIDNum: 10000009
  896. CallerIDName:
  897. Uniqueid: 1296049342.11
  898.  
  899.  
  900. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  901. Event: VarSet
  902. Privilege: dialplan,all
  903. Channel: SIP/dovid-0000000a
  904. Variable: DIALSTATUS
  905. Value: ANSWER
  906. Uniqueid: 1296049342.10
  907.  
  908.  
  909. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  910. Event: VarSet
  911. Privilege: dialplan,all
  912. Channel: SIP/dovid-0000000a
  913. Variable: DIALEDPEERNAME
  914. Value: SIP/fpp-0000000b
  915. Uniqueid: 1296049342.10
  916.  
  917.  
  918. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  919. Event: VarSet
  920. Privilege: dialplan,all
  921. Channel: SIP/dovid-0000000a
  922. Variable: DIALEDPEERNUMBER
  923. Value: 10000009@fpp
  924. Uniqueid: 1296049342.10
  925.  
  926.  
  927. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  928. Event: VarSet
  929. Privilege: dialplan,all
  930. Channel: SIP/fpp-0000000b
  931. Variable: LOCAL(ARG1)
  932. Value: s
  933. Uniqueid: 1296049342.11
  934.  
  935.  
  936. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  937. Event: VarSet
  938. Privilege: dialplan,all
  939. Channel: SIP/fpp-0000000b
  940. Variable: LOCAL(ARG2)
  941. Value: 1
  942. Uniqueid: 1296049342.11
  943.  
  944.  
  945. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  946. Event: VarSet
  947. Privilege: dialplan,all
  948. Channel: SIP/fpp-0000000b
  949. Variable: LOCAL(ARGC)
  950. Value: 2
  951. Uniqueid: 1296049342.11
  952.  
  953.  
  954. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  955. Event: Newexten
  956. Privilege: dialplan,all
  957. Channel: SIP/fpp-0000000b
  958. Context: do_dtmf_cc-take-call
  959. Extension: s
  960. Priority: 1
  961. Application: AGI
  962. AppData: agi://127.0.0.1:4579/update_call_status?status=60
  963. Uniqueid: 1296049342.11
  964.  
  965.  
  966. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  967. Event: AGIExec
  968. Privilege: agi,all
  969. SubEvent: Start
  970. Channel: SIP/fpp-0000000b
  971. CommandId: 618917567
  972. Command: GET VARIABLE our_start
  973.  
  974.  
  975. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'our_start' is NULL
  976. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  977. Event: AGIExec
  978. Privilege: agi,all
  979. SubEvent: End
  980. Channel: SIP/fpp-0000000b
  981. CommandId: 618917567
  982. Command: GET VARIABLE our_start
  983. ResultCode: 200
  984. Result: Success
  985.  
  986.  
  987. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'uuid' is NULL
  988. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  989. Event: AGIExec
  990. Privilege: agi,all
  991. SubEvent: Start
  992. Channel: SIP/fpp-0000000b
  993. CommandId: 1431956463
  994. Command: GET VARIABLE uuid
  995.  
  996.  
  997. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  998. Event: AGIExec
  999. Privilege: agi,all
  1000. SubEvent: End
  1001. Channel: SIP/fpp-0000000b
  1002. CommandId: 1431956463
  1003. Command: GET VARIABLE uuid
  1004. ResultCode: 200
  1005. Result: Success
  1006.  
  1007.  
  1008. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'recording' is NULL
  1009. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1010. Event: AGIExec
  1011. Privilege: agi,all
  1012. SubEvent: Start
  1013. Channel: SIP/fpp-0000000b
  1014. CommandId: 841530017
  1015. Command: GET VARIABLE recording
  1016.  
  1017.  
  1018. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1019. Event: AGIExec
  1020. Privilege: agi,all
  1021. SubEvent: End
  1022. Channel: SIP/fpp-0000000b
  1023. CommandId: 841530017
  1024. Command: GET VARIABLE recording
  1025. ResultCode: 200
  1026. Result: Success
  1027.  
  1028.  
  1029. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'rec_file' is NULL
  1030. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1031. Event: AGIExec
  1032. Privilege: agi,all
  1033. SubEvent: Start
  1034. Channel: SIP/fpp-0000000b
  1035. CommandId: 94806851
  1036. Command: GET VARIABLE rec_file
  1037.  
  1038.  
  1039. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1040. Event: AGIExec
  1041. Privilege: agi,all
  1042. SubEvent: End
  1043. Channel: SIP/fpp-0000000b
  1044. CommandId: 94806851
  1045. Command: GET VARIABLE rec_file
  1046. ResultCode: 200
  1047. Result: Success
  1048.  
  1049.  
  1050. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'pass' is NULL
  1051. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1052. Event: AGIExec
  1053. Privilege: agi,all
  1054. SubEvent: Start
  1055. Channel: SIP/fpp-0000000b
  1056. CommandId: 1967464179
  1057. Command: GET VARIABLE pass
  1058.  
  1059.  
  1060. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1061. Event: AGIExec
  1062. Privilege: agi,all
  1063. SubEvent: End
  1064. Channel: SIP/fpp-0000000b
  1065. CommandId: 1967464179
  1066. Command: GET VARIABLE pass
  1067. ResultCode: 200
  1068. Result: Success
  1069.  
  1070.  
  1071. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'lega' is NULL
  1072. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1073. Event: AGIExec
  1074. Privilege: agi,all
  1075. SubEvent: Start
  1076. Channel: SIP/fpp-0000000b
  1077. CommandId: 1573584841
  1078. Command: GET VARIABLE lega
  1079.  
  1080.  
  1081. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1082. Event: AGIExec
  1083. Privilege: agi,all
  1084. SubEvent: End
  1085. Channel: SIP/fpp-0000000b
  1086. CommandId: 1573584841
  1087. Command: GET VARIABLE lega
  1088. ResultCode: 200
  1089. Result: Success
  1090.  
  1091.  
  1092. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'legb' is NULL
  1093. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1094. Event: AGIExec
  1095. Privilege: agi,all
  1096. SubEvent: Start
  1097. Channel: SIP/fpp-0000000b
  1098. CommandId: 708202332
  1099. Command: GET VARIABLE legb
  1100.  
  1101.  
  1102. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1103. Event: AGIExec
  1104. Privilege: agi,all
  1105. SubEvent: End
  1106. Channel: SIP/fpp-0000000b
  1107. CommandId: 708202332
  1108. Command: GET VARIABLE legb
  1109. ResultCode: 200
  1110. Result: Success
  1111.  
  1112.  
  1113. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'cida' is NULL
  1114. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1115. Event: AGIExec
  1116. Privilege: agi,all
  1117. SubEvent: Start
  1118. Channel: SIP/fpp-0000000b
  1119. CommandId: 990087855
  1120. Command: GET VARIABLE cida
  1121.  
  1122.  
  1123. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1124. Event: AGIExec
  1125. Privilege: agi,all
  1126. SubEvent: End
  1127. Channel: SIP/fpp-0000000b
  1128. CommandId: 990087855
  1129. Command: GET VARIABLE cida
  1130. ResultCode: 200
  1131. Result: Success
  1132.  
  1133.  
  1134. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'cidb' is NULL
  1135. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1136. Event: AGIExec
  1137. Privilege: agi,all
  1138. SubEvent: Start
  1139. Channel: SIP/fpp-0000000b
  1140. CommandId: 1949468822
  1141. Command: GET VARIABLE cidb
  1142.  
  1143.  
  1144. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1145. Event: AGIExec
  1146. Privilege: agi,all
  1147. SubEvent: End
  1148. Channel: SIP/fpp-0000000b
  1149. CommandId: 1949468822
  1150. Command: GET VARIABLE cidb
  1151. ResultCode: 200
  1152. Result: Success
  1153.  
  1154.  
  1155. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'send_dtmf' is NULL
  1156. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1157. Event: AGIExec
  1158. Privilege: agi,all
  1159. SubEvent: Start
  1160. Channel: SIP/fpp-0000000b
  1161. CommandId: 1400571379
  1162. Command: GET VARIABLE send_dtmf
  1163.  
  1164.  
  1165. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1166. Event: AGIExec
  1167. Privilege: agi,all
  1168. SubEvent: End
  1169. Channel: SIP/fpp-0000000b
  1170. CommandId: 1400571379
  1171. Command: GET VARIABLE send_dtmf
  1172. ResultCode: 200
  1173. Result: Success
  1174.  
  1175.  
  1176. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'dtmf' is NULL
  1177. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1178. Event: AGIExec
  1179. Privilege: agi,all
  1180. SubEvent: Start
  1181. Channel: SIP/fpp-0000000b
  1182. CommandId: 206538877
  1183. Command: GET VARIABLE dtmf
  1184.  
  1185.  
  1186. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1187. Event: AGIExec
  1188. Privilege: agi,all
  1189. SubEvent: End
  1190. Channel: SIP/fpp-0000000b
  1191. CommandId: 206538877
  1192. Command: GET VARIABLE dtmf
  1193. ResultCode: 200
  1194. Result: Success
  1195.  
  1196.  
  1197. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'wava1' is NULL
  1198. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1199. Event: AGIExec
  1200. Privilege: agi,all
  1201. SubEvent: Start
  1202. Channel: SIP/fpp-0000000b
  1203. CommandId: 2026532005
  1204. Command: GET VARIABLE wava1
  1205.  
  1206.  
  1207. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1208. Event: AGIExec
  1209. Privilege: agi,all
  1210. SubEvent: End
  1211. Channel: SIP/fpp-0000000b
  1212. CommandId: 2026532005
  1213. Command: GET VARIABLE wava1
  1214. ResultCode: 200
  1215. Result: Success
  1216.  
  1217.  
  1218. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'play_wava2' is NULL
  1219. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1220. Event: AGIExec
  1221. Privilege: agi,all
  1222. SubEvent: Start
  1223. Channel: SIP/fpp-0000000b
  1224. CommandId: 67556870
  1225. Command: GET VARIABLE play_wava2
  1226.  
  1227.  
  1228. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1229. Event: AGIExec
  1230. Privilege: agi,all
  1231. SubEvent: End
  1232. Channel: SIP/fpp-0000000b
  1233. CommandId: 67556870
  1234. Command: GET VARIABLE play_wava2
  1235. ResultCode: 200
  1236. Result: Success
  1237.  
  1238.  
  1239. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'wava2' is NULL
  1240. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1241. Event: AGIExec
  1242. Privilege: agi,all
  1243. SubEvent: Start
  1244. Channel: SIP/fpp-0000000b
  1245. CommandId: 2003954683
  1246. Command: GET VARIABLE wava2
  1247.  
  1248.  
  1249. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1250. Event: AGIExec
  1251. Privilege: agi,all
  1252. SubEvent: End
  1253. Channel: SIP/fpp-0000000b
  1254. CommandId: 2003954683
  1255. Command: GET VARIABLE wava2
  1256. ResultCode: 200
  1257. Result: Success
  1258.  
  1259.  
  1260. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'play_wavb' is NULL
  1261. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1262. Event: AGIExec
  1263. Privilege: agi,all
  1264. SubEvent: Start
  1265. Channel: SIP/fpp-0000000b
  1266. CommandId: 1213119484
  1267. Command: GET VARIABLE play_wavb
  1268.  
  1269.  
  1270. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1271. Event: AGIExec
  1272. Privilege: agi,all
  1273. SubEvent: End
  1274. Channel: SIP/fpp-0000000b
  1275. CommandId: 1213119484
  1276. Command: GET VARIABLE play_wavb
  1277. ResultCode: 200
  1278. Result: Success
  1279.  
  1280.  
  1281. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'wavb' is NULL
  1282. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1283. Event: AGIExec
  1284. Privilege: agi,all
  1285. SubEvent: Start
  1286. Channel: SIP/fpp-0000000b
  1287. CommandId: 1978031554
  1288. Command: GET VARIABLE wavb
  1289.  
  1290.  
  1291. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1292. Event: AGIExec
  1293. Privilege: agi,all
  1294. SubEvent: End
  1295. Channel: SIP/fpp-0000000b
  1296. CommandId: 1978031554
  1297. Command: GET VARIABLE wavb
  1298. ResultCode: 200
  1299. Result: Success
  1300.  
  1301.  
  1302. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'play_wava3' is NULL
  1303. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1304. Event: AGIExec
  1305. Privilege: agi,all
  1306. SubEvent: Start
  1307. Channel: SIP/fpp-0000000b
  1308. CommandId: 1583833265
  1309. Command: GET VARIABLE play_wava3
  1310.  
  1311.  
  1312. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1313. Event: AGIExec
  1314. Privilege: agi,all
  1315. SubEvent: End
  1316. Channel: SIP/fpp-0000000b
  1317. CommandId: 1583833265
  1318. Command: GET VARIABLE play_wava3
  1319. ResultCode: 200
  1320. Result: Success
  1321.  
  1322.  
  1323. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'wava3' is NULL
  1324. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1325. Event: AGIExec
  1326. Privilege: agi,all
  1327. SubEvent: Start
  1328. Channel: SIP/fpp-0000000b
  1329. CommandId: 869225679
  1330. Command: GET VARIABLE wava3
  1331.  
  1332.  
  1333. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1334. Event: AGIExec
  1335. Privilege: agi,all
  1336. SubEvent: End
  1337. Channel: SIP/fpp-0000000b
  1338. CommandId: 869225679
  1339. Command: GET VARIABLE wava3
  1340. ResultCode: 200
  1341. Result: Success
  1342.  
  1343.  
  1344. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Result of 'timeout' is NULL
  1345. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1346. Event: AGIExec
  1347. Privilege: agi,all
  1348. SubEvent: Start
  1349. Channel: SIP/fpp-0000000b
  1350. CommandId: 145917225
  1351. Command: GET VARIABLE timeout
  1352.  
  1353.  
  1354. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1355. Event: AGIExec
  1356. Privilege: agi,all
  1357. SubEvent: End
  1358. Channel: SIP/fpp-0000000b
  1359. CommandId: 145917225
  1360. Command: GET VARIABLE timeout
  1361. ResultCode: 200
  1362. Result: Success
  1363.  
  1364.  
  1365. [Jan 26 08:42:25] VERBOSE[23821] res_agi.c: -- <SIP/fpp-0000000b>AGI Script agi://127.0.0.1:4579/update_call_status?status=60 completed, returning 0
  1366. [Jan 26 08:42:25] DEBUG[23821] pbx.c: Launching 'SendDTMF'
  1367. [Jan 26 08:42:25] VERBOSE[23821] pbx.c: -- Executing [s@do_dtmf_cc-take-call:2] SendDTMF("SIP/fpp-0000000b", "123456") in new stack
  1368. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1369. Event: VarSet
  1370. Privilege: dialplan,all
  1371. Channel: SIP/fpp-0000000b
  1372. Variable: AGISTATUS
  1373. Value: SUCCESS
  1374. Uniqueid: 1296049342.11
  1375.  
  1376.  
  1377. [Jan 26 08:42:25] DEBUG[23750] manager.c: Examining event:
  1378. Event: Newexten
  1379. Privilege: dialplan,all
  1380. Channel: SIP/fpp-0000000b
  1381. Context: do_dtmf_cc-take-call
  1382. Extension: s
  1383. Priority: 2
  1384. Application: SendDTMF
  1385. AppData: 123456
  1386. Uniqueid: 1296049342.11
  1387.  
  1388.  
  1389. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Launching 'AGI'
  1390. [Jan 26 08:42:27] VERBOSE[23821] pbx.c: -- Executing [s@do_dtmf_cc-take-call:3] AGI("SIP/fpp-0000000b", "agi://127.0.0.1:4579/update_call_status?status=70") in new stack
  1391. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1392. Event: Newexten
  1393. Privilege: dialplan,all
  1394. Channel: SIP/fpp-0000000b
  1395. Context: do_dtmf_cc-take-call
  1396. Extension: s
  1397. Priority: 3
  1398. Application: AGI
  1399. AppData: agi://127.0.0.1:4579/update_call_status?status=70
  1400. Uniqueid: 1296049342.11
  1401.  
  1402.  
  1403. [Jan 26 08:42:27] DEBUG[23821] res_agi.c: Wow, connected!
  1404. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1405. Event: AGIExec
  1406. Privilege: agi,all
  1407. SubEvent: Start
  1408. Channel: SIP/fpp-0000000b
  1409. CommandId: 660440399
  1410. Command: GET VARIABLE our_start
  1411.  
  1412.  
  1413. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'our_start' is NULL
  1414. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1415. Event: AGIExec
  1416. Privilege: agi,all
  1417. SubEvent: End
  1418. Channel: SIP/fpp-0000000b
  1419. CommandId: 660440399
  1420. Command: GET VARIABLE our_start
  1421. ResultCode: 200
  1422. Result: Success
  1423.  
  1424.  
  1425. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'uuid' is NULL
  1426. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1427. Event: AGIExec
  1428. Privilege: agi,all
  1429. SubEvent: Start
  1430. Channel: SIP/fpp-0000000b
  1431. CommandId: 1496595469
  1432. Command: GET VARIABLE uuid
  1433.  
  1434.  
  1435. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1436. Event: AGIExec
  1437. Privilege: agi,all
  1438. SubEvent: End
  1439. Channel: SIP/fpp-0000000b
  1440. CommandId: 1496595469
  1441. Command: GET VARIABLE uuid
  1442. ResultCode: 200
  1443. Result: Success
  1444.  
  1445.  
  1446. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'recording' is NULL
  1447. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1448. Event: AGIExec
  1449. Privilege: agi,all
  1450. SubEvent: Start
  1451. Channel: SIP/fpp-0000000b
  1452. CommandId: 399081823
  1453. Command: GET VARIABLE recording
  1454.  
  1455.  
  1456. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1457. Event: AGIExec
  1458. Privilege: agi,all
  1459. SubEvent: End
  1460. Channel: SIP/fpp-0000000b
  1461. CommandId: 399081823
  1462. Command: GET VARIABLE recording
  1463. ResultCode: 200
  1464. Result: Success
  1465.  
  1466.  
  1467. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'rec_file' is NULL
  1468. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1469. Event: AGIExec
  1470. Privilege: agi,all
  1471. SubEvent: Start
  1472. Channel: SIP/fpp-0000000b
  1473. CommandId: 1515363297
  1474. Command: GET VARIABLE rec_file
  1475.  
  1476.  
  1477. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1478. Event: AGIExec
  1479. Privilege: agi,all
  1480. SubEvent: End
  1481. Channel: SIP/fpp-0000000b
  1482. CommandId: 1515363297
  1483. Command: GET VARIABLE rec_file
  1484. ResultCode: 200
  1485. Result: Success
  1486.  
  1487.  
  1488. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'pass' is NULL
  1489. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1490. Event: AGIExec
  1491. Privilege: agi,all
  1492. SubEvent: Start
  1493. Channel: SIP/fpp-0000000b
  1494. CommandId: 1515893988
  1495. Command: GET VARIABLE pass
  1496.  
  1497.  
  1498. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1499. Event: AGIExec
  1500. Privilege: agi,all
  1501. SubEvent: End
  1502. Channel: SIP/fpp-0000000b
  1503. CommandId: 1515893988
  1504. Command: GET VARIABLE pass
  1505. ResultCode: 200
  1506. Result: Success
  1507.  
  1508.  
  1509. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'lega' is NULL
  1510. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1511. Event: AGIExec
  1512. Privilege: agi,all
  1513. SubEvent: Start
  1514. Channel: SIP/fpp-0000000b
  1515. CommandId: 1264369181
  1516. Command: GET VARIABLE lega
  1517.  
  1518.  
  1519. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1520. Event: AGIExec
  1521. Privilege: agi,all
  1522. SubEvent: End
  1523. Channel: SIP/fpp-0000000b
  1524. CommandId: 1264369181
  1525. Command: GET VARIABLE lega
  1526. ResultCode: 200
  1527. Result: Success
  1528.  
  1529.  
  1530. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'legb' is NULL
  1531. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1532. Event: AGIExec
  1533. Privilege: agi,all
  1534. SubEvent: Start
  1535. Channel: SIP/fpp-0000000b
  1536. CommandId: 1385029579
  1537. Command: GET VARIABLE legb
  1538.  
  1539.  
  1540. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1541. Event: AGIExec
  1542. Privilege: agi,all
  1543. SubEvent: End
  1544. Channel: SIP/fpp-0000000b
  1545. CommandId: 1385029579
  1546. Command: GET VARIABLE legb
  1547. ResultCode: 200
  1548. Result: Success
  1549.  
  1550.  
  1551. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'cida' is NULL
  1552. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1553. Event: AGIExec
  1554. Privilege: agi,all
  1555. SubEvent: Start
  1556. Channel: SIP/fpp-0000000b
  1557. CommandId: 177557854
  1558. Command: GET VARIABLE cida
  1559.  
  1560.  
  1561. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1562. Event: AGIExec
  1563. Privilege: agi,all
  1564. SubEvent: End
  1565. Channel: SIP/fpp-0000000b
  1566. CommandId: 177557854
  1567. Command: GET VARIABLE cida
  1568. ResultCode: 200
  1569. Result: Success
  1570.  
  1571.  
  1572. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'cidb' is NULL
  1573. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1574. Event: AGIExec
  1575. Privilege: agi,all
  1576. SubEvent: Start
  1577. Channel: SIP/fpp-0000000b
  1578. CommandId: 12443146
  1579. Command: GET VARIABLE cidb
  1580.  
  1581.  
  1582. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1583. Event: AGIExec
  1584. Privilege: agi,all
  1585. SubEvent: End
  1586. Channel: SIP/fpp-0000000b
  1587. CommandId: 12443146
  1588. Command: GET VARIABLE cidb
  1589. ResultCode: 200
  1590. Result: Success
  1591.  
  1592.  
  1593. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'send_dtmf' is NULL
  1594. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1595. Event: AGIExec
  1596. Privilege: agi,all
  1597. SubEvent: Start
  1598. Channel: SIP/fpp-0000000b
  1599. CommandId: 1707848121
  1600. Command: GET VARIABLE send_dtmf
  1601.  
  1602.  
  1603. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1604. Event: AGIExec
  1605. Privilege: agi,all
  1606. SubEvent: End
  1607. Channel: SIP/fpp-0000000b
  1608. CommandId: 1707848121
  1609. Command: GET VARIABLE send_dtmf
  1610. ResultCode: 200
  1611. Result: Success
  1612.  
  1613.  
  1614. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'dtmf' is NULL
  1615. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1616. Event: AGIExec
  1617. Privilege: agi,all
  1618. SubEvent: Start
  1619. Channel: SIP/fpp-0000000b
  1620. CommandId: 1099702212
  1621. Command: GET VARIABLE dtmf
  1622.  
  1623.  
  1624. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1625. Event: AGIExec
  1626. Privilege: agi,all
  1627. SubEvent: End
  1628. Channel: SIP/fpp-0000000b
  1629. CommandId: 1099702212
  1630. Command: GET VARIABLE dtmf
  1631. ResultCode: 200
  1632. Result: Success
  1633.  
  1634.  
  1635. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'wava1' is NULL
  1636. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1637. Event: AGIExec
  1638. Privilege: agi,all
  1639. SubEvent: Start
  1640. Channel: SIP/fpp-0000000b
  1641. CommandId: 928163237
  1642. Command: GET VARIABLE wava1
  1643.  
  1644.  
  1645. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1646. Event: AGIExec
  1647. Privilege: agi,all
  1648. SubEvent: End
  1649. Channel: SIP/fpp-0000000b
  1650. CommandId: 928163237
  1651. Command: GET VARIABLE wava1
  1652. ResultCode: 200
  1653. Result: Success
  1654.  
  1655.  
  1656. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'play_wava2' is NULL
  1657. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1658. Event: AGIExec
  1659. Privilege: agi,all
  1660. SubEvent: Start
  1661. Channel: SIP/fpp-0000000b
  1662. CommandId: 880758580
  1663. Command: GET VARIABLE play_wava2
  1664.  
  1665.  
  1666. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1667. Event: AGIExec
  1668. Privilege: agi,all
  1669. SubEvent: End
  1670. Channel: SIP/fpp-0000000b
  1671. CommandId: 880758580
  1672. Command: GET VARIABLE play_wava2
  1673. ResultCode: 200
  1674. Result: Success
  1675.  
  1676.  
  1677. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'wava2' is NULL
  1678. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1679. Event: AGIExec
  1680. Privilege: agi,all
  1681. SubEvent: Start
  1682. Channel: SIP/fpp-0000000b
  1683. CommandId: 85549054
  1684. Command: GET VARIABLE wava2
  1685.  
  1686.  
  1687. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1688. Event: AGIExec
  1689. Privilege: agi,all
  1690. SubEvent: End
  1691. Channel: SIP/fpp-0000000b
  1692. CommandId: 85549054
  1693. Command: GET VARIABLE wava2
  1694. ResultCode: 200
  1695. Result: Success
  1696.  
  1697.  
  1698. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'play_wavb' is NULL
  1699. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1700. Event: AGIExec
  1701. Privilege: agi,all
  1702. SubEvent: Start
  1703. Channel: SIP/fpp-0000000b
  1704. CommandId: 1838069862
  1705. Command: GET VARIABLE play_wavb
  1706.  
  1707.  
  1708. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1709. Event: AGIExec
  1710. Privilege: agi,all
  1711. SubEvent: End
  1712. Channel: SIP/fpp-0000000b
  1713. CommandId: 1838069862
  1714. Command: GET VARIABLE play_wavb
  1715. ResultCode: 200
  1716. Result: Success
  1717.  
  1718.  
  1719. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'wavb' is NULL
  1720. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1721. Event: AGIExec
  1722. Privilege: agi,all
  1723. SubEvent: Start
  1724. Channel: SIP/fpp-0000000b
  1725. CommandId: 1493295641
  1726. Command: GET VARIABLE wavb
  1727.  
  1728.  
  1729. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1730. Event: AGIExec
  1731. Privilege: agi,all
  1732. SubEvent: End
  1733. Channel: SIP/fpp-0000000b
  1734. CommandId: 1493295641
  1735. Command: GET VARIABLE wavb
  1736. ResultCode: 200
  1737. Result: Success
  1738.  
  1739.  
  1740. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'play_wava3' is NULL
  1741. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1742. Event: AGIExec
  1743. Privilege: agi,all
  1744. SubEvent: Start
  1745. Channel: SIP/fpp-0000000b
  1746. CommandId: 1836577048
  1747. Command: GET VARIABLE play_wava3
  1748.  
  1749.  
  1750. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1751. Event: AGIExec
  1752. Privilege: agi,all
  1753. SubEvent: End
  1754. Channel: SIP/fpp-0000000b
  1755. CommandId: 1836577048
  1756. Command: GET VARIABLE play_wava3
  1757. ResultCode: 200
  1758. Result: Success
  1759.  
  1760.  
  1761. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'wava3' is NULL
  1762. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1763. Event: AGIExec
  1764. Privilege: agi,all
  1765. SubEvent: Start
  1766. Channel: SIP/fpp-0000000b
  1767. CommandId: 2064680356
  1768. Command: GET VARIABLE wava3
  1769.  
  1770.  
  1771. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1772. Event: AGIExec
  1773. Privilege: agi,all
  1774. SubEvent: End
  1775. Channel: SIP/fpp-0000000b
  1776. CommandId: 2064680356
  1777. Command: GET VARIABLE wava3
  1778. ResultCode: 200
  1779. Result: Success
  1780.  
  1781.  
  1782. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'timeout' is NULL
  1783. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1784. Event: AGIExec
  1785. Privilege: agi,all
  1786. SubEvent: Start
  1787. Channel: SIP/fpp-0000000b
  1788. CommandId: 442506008
  1789. Command: GET VARIABLE timeout
  1790.  
  1791.  
  1792. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1793. Event: AGIExec
  1794. Privilege: agi,all
  1795. SubEvent: End
  1796. Channel: SIP/fpp-0000000b
  1797. CommandId: 442506008
  1798. Command: GET VARIABLE timeout
  1799. ResultCode: 200
  1800. Result: Success
  1801.  
  1802.  
  1803. [Jan 26 08:42:27] VERBOSE[23821] res_agi.c: -- <SIP/fpp-0000000b>AGI Script agi://127.0.0.1:4579/update_call_status?status=70 completed, returning 0
  1804. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Result of 'EPOCH' is '1296049347'
  1805. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Launching 'Set'
  1806. [Jan 26 08:42:27] VERBOSE[23821] pbx.c: -- Executing [s@do_dtmf_cc-take-call:4] Set("SIP/fpp-0000000b", "wavb_start=1296049347") in new stack
  1807. [Jan 26 08:42:27] DEBUG[23821] pbx.c: Launching 'BackGround'
  1808. [Jan 26 08:42:27] VERBOSE[23821] pbx.c: -- Executing [s@do_dtmf_cc-take-call:5] BackGround("SIP/fpp-0000000b", "/etc/cb/wav/incoming_cb_call") in new stack
  1809. [Jan 26 08:42:27] DEBUG[23821] channel.c: Set channel SIP/fpp-0000000b to write format gsm
  1810. [Jan 26 08:42:27] DEBUG[23821] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
  1811. [Jan 26 08:42:27] DEBUG[23821] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
  1812. [Jan 26 08:42:27] DEBUG[23821] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  1813. [Jan 26 08:42:27] VERBOSE[23821] file.c: -- <SIP/fpp-0000000b> Playing '/etc/cb/wav/incoming_cb_call.gsm' (language 'en')
  1814. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1815. Event: VarSet
  1816. Privilege: dialplan,all
  1817. Channel: SIP/fpp-0000000b
  1818. Variable: AGISTATUS
  1819. Value: SUCCESS
  1820. Uniqueid: 1296049342.11
  1821.  
  1822.  
  1823. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1824. Event: Newexten
  1825. Privilege: dialplan,all
  1826. Channel: SIP/fpp-0000000b
  1827. Context: do_dtmf_cc-take-call
  1828. Extension: s
  1829. Priority: 4
  1830. Application: Set
  1831. AppData: wavb_start=1296049347
  1832. Uniqueid: 1296049342.11
  1833.  
  1834.  
  1835. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1836. Event: VarSet
  1837. Privilege: dialplan,all
  1838. Channel: SIP/fpp-0000000b
  1839. Variable: wavb_start
  1840. Value: 1296049347
  1841. Uniqueid: 1296049342.11
  1842.  
  1843.  
  1844. [Jan 26 08:42:27] DEBUG[23750] manager.c: Examining event:
  1845. Event: Newexten
  1846. Privilege: dialplan,all
  1847. Channel: SIP/fpp-0000000b
  1848. Context: do_dtmf_cc-take-call
  1849. Extension: s
  1850. Priority: 5
  1851. Application: BackGround
  1852. AppData: /etc/cb/wav/incoming_cb_call
  1853. Uniqueid: 1296049342.11
  1854.  
  1855.  
  1856. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1857. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Sending dtmf: 49 (1), at 69.167.68.133:15760
  1858. [Jan 26 08:42:30] DTMF[23821] channel.c: DTMF begin '1' received on SIP/fpp-0000000b
  1859. [Jan 26 08:42:30] DTMF[23821] channel.c: DTMF begin ignored '1' on SIP/fpp-0000000b
  1860. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1861. Event: DTMF
  1862. Privilege: dtmf,all
  1863. Channel: SIP/fpp-0000000b
  1864. Uniqueid: 1296049342.11
  1865. Digit: 1
  1866. Direction: Received
  1867. Begin: Yes
  1868. End: No
  1869.  
  1870.  
  1871. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1872. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1873. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1874. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 64 bytes
  1875. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1876. Event: RTCPReceived
  1877. Privilege: reporting,all
  1878. From 69.167.68.133:15761
  1879. PT: 200(Sender Report)
  1880. ReceptionReports: 1
  1881. SenderSSRC: 0
  1882. FractionLost: 0
  1883. PacketsLost: 0
  1884. HighestSequence: 23422
  1885. SequenceNumberCycles: 0
  1886. IAJitter: 6
  1887. LastSR: 0.0000000000
  1888. DLSR: 9414.3980(sec)
  1889.  
  1890.  
  1891. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1892. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1893. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1894. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1895. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Sending dtmf: 49 (1), at 69.167.68.133:15760
  1896. [Jan 26 08:42:30] DTMF[23821] channel.c: DTMF end '1' received on SIP/fpp-0000000b, duration 160 ms
  1897. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1898. Event: DTMF
  1899. Privilege: dtmf,all
  1900. Channel: SIP/fpp-0000000b
  1901. Uniqueid: 1296049342.11
  1902. Digit: 1
  1903. Direction: Received
  1904. Begin: No
  1905. End: Yes
  1906.  
  1907.  
  1908. [Jan 26 08:42:30] DTMF[23821] channel.c: DTMF end passthrough '1' on SIP/fpp-0000000b
  1909. [Jan 26 08:42:30] DEBUG[23821] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  1910. [Jan 26 08:42:30] DEBUG[23821] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  1911. [Jan 26 08:42:30] DEBUG[23821] channel.c: Set channel SIP/fpp-0000000b to write format ulaw
  1912. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Launching 'AGI'
  1913. [Jan 26 08:42:30] VERBOSE[23821] pbx.c: -- Executing [1@do_dtmf_cc-take-call:1] AGI("SIP/fpp-0000000b", "agi://127.0.0.1:4579/update_call_status?status=80") in new stack
  1914. [Jan 26 08:42:30] DEBUG[23821] res_agi.c: Wow, connected!
  1915. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1916. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4)
  1917. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1918. Event: VarSet
  1919. Privilege: dialplan,all
  1920. Channel: SIP/fpp-0000000b
  1921. Variable: BACKGROUNDSTATUS
  1922. Value: SUCCESS
  1923. Uniqueid: 1296049342.11
  1924.  
  1925.  
  1926. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1927. Event: Newexten
  1928. Privilege: dialplan,all
  1929. Channel: SIP/fpp-0000000b
  1930. Context: do_dtmf_cc-take-call
  1931. Extension: 1
  1932. Priority: 1
  1933. Application: AGI
  1934. AppData: agi://127.0.0.1:4579/update_call_status?status=80
  1935. Uniqueid: 1296049342.11
  1936.  
  1937.  
  1938. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1939. Event: RTCPSent
  1940. Privilege: reporting,all
  1941. To 69.167.68.133:15761
  1942. OurSSRC: 529697820
  1943. SentNTP: 1296049350.1563152384
  1944. SentRTP: 29760
  1945. SentPackets: 141
  1946. SentOctets: 22560
  1947. ReportBlock:
  1948. FractionLost: 0
  1949. CumulativeLoss: 0
  1950. IAJitter: 0.0028
  1951. TheirLastSR: 2739299819
  1952. DLSR: 0.0700 (sec)
  1953.  
  1954.  
  1955. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1956. Event: AGIExec
  1957. Privilege: agi,all
  1958. SubEvent: Start
  1959. Channel: SIP/fpp-0000000b
  1960. CommandId: 1982505854
  1961. Command: GET VARIABLE our_start
  1962.  
  1963.  
  1964. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'our_start' is NULL
  1965. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1966. Event: AGIExec
  1967. Privilege: agi,all
  1968. SubEvent: End
  1969. Channel: SIP/fpp-0000000b
  1970. CommandId: 1982505854
  1971. Command: GET VARIABLE our_start
  1972. ResultCode: 200
  1973. Result: Success
  1974.  
  1975.  
  1976. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'uuid' is NULL
  1977. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1978. Event: AGIExec
  1979. Privilege: agi,all
  1980. SubEvent: Start
  1981. Channel: SIP/fpp-0000000b
  1982. CommandId: 1885220917
  1983. Command: GET VARIABLE uuid
  1984.  
  1985.  
  1986. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1987. Event: AGIExec
  1988. Privilege: agi,all
  1989. SubEvent: End
  1990. Channel: SIP/fpp-0000000b
  1991. CommandId: 1885220917
  1992. Command: GET VARIABLE uuid
  1993. ResultCode: 200
  1994. Result: Success
  1995.  
  1996.  
  1997. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'recording' is NULL
  1998. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  1999. Event: AGIExec
  2000. Privilege: agi,all
  2001. SubEvent: Start
  2002. Channel: SIP/fpp-0000000b
  2003. CommandId: 1567842072
  2004. Command: GET VARIABLE recording
  2005.  
  2006.  
  2007. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2008. Event: AGIExec
  2009. Privilege: agi,all
  2010. SubEvent: End
  2011. Channel: SIP/fpp-0000000b
  2012. CommandId: 1567842072
  2013. Command: GET VARIABLE recording
  2014. ResultCode: 200
  2015. Result: Success
  2016.  
  2017.  
  2018. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'rec_file' is NULL
  2019. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2020. Event: AGIExec
  2021. Privilege: agi,all
  2022. SubEvent: Start
  2023. Channel: SIP/fpp-0000000b
  2024. CommandId: 1318386563
  2025. Command: GET VARIABLE rec_file
  2026.  
  2027.  
  2028. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2029. Event: AGIExec
  2030. Privilege: agi,all
  2031. SubEvent: End
  2032. Channel: SIP/fpp-0000000b
  2033. CommandId: 1318386563
  2034. Command: GET VARIABLE rec_file
  2035. ResultCode: 200
  2036. Result: Success
  2037.  
  2038.  
  2039. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'pass' is NULL
  2040. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2041. Event: AGIExec
  2042. Privilege: agi,all
  2043. SubEvent: Start
  2044. Channel: SIP/fpp-0000000b
  2045. CommandId: 1178288907
  2046. Command: GET VARIABLE pass
  2047.  
  2048.  
  2049. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2050. Event: AGIExec
  2051. Privilege: agi,all
  2052. SubEvent: End
  2053. Channel: SIP/fpp-0000000b
  2054. CommandId: 1178288907
  2055. Command: GET VARIABLE pass
  2056. ResultCode: 200
  2057. Result: Success
  2058.  
  2059.  
  2060. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'lega' is NULL
  2061. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2062. Event: AGIExec
  2063. Privilege: agi,all
  2064. SubEvent: Start
  2065. Channel: SIP/fpp-0000000b
  2066. CommandId: 118399429
  2067. Command: GET VARIABLE lega
  2068.  
  2069.  
  2070. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2071. Event: AGIExec
  2072. Privilege: agi,all
  2073. SubEvent: End
  2074. Channel: SIP/fpp-0000000b
  2075. CommandId: 118399429
  2076. Command: GET VARIABLE lega
  2077. ResultCode: 200
  2078. Result: Success
  2079.  
  2080.  
  2081. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'legb' is NULL
  2082. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2083. Event: AGIExec
  2084. Privilege: agi,all
  2085. SubEvent: Start
  2086. Channel: SIP/fpp-0000000b
  2087. CommandId: 1923776214
  2088. Command: GET VARIABLE legb
  2089.  
  2090.  
  2091. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2092. Event: AGIExec
  2093. Privilege: agi,all
  2094. SubEvent: End
  2095. Channel: SIP/fpp-0000000b
  2096. CommandId: 1923776214
  2097. Command: GET VARIABLE legb
  2098. ResultCode: 200
  2099. Result: Success
  2100.  
  2101.  
  2102. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'cida' is NULL
  2103. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2104. Event: AGIExec
  2105. Privilege: agi,all
  2106. SubEvent: Start
  2107. Channel: SIP/fpp-0000000b
  2108. CommandId: 1408410146
  2109. Command: GET VARIABLE cida
  2110.  
  2111.  
  2112. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2113. Event: AGIExec
  2114. Privilege: agi,all
  2115. SubEvent: End
  2116. Channel: SIP/fpp-0000000b
  2117. CommandId: 1408410146
  2118. Command: GET VARIABLE cida
  2119. ResultCode: 200
  2120. Result: Success
  2121.  
  2122.  
  2123. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'cidb' is NULL
  2124. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2125. Event: AGIExec
  2126. Privilege: agi,all
  2127. SubEvent: Start
  2128. Channel: SIP/fpp-0000000b
  2129. CommandId: 1055805639
  2130. Command: GET VARIABLE cidb
  2131.  
  2132.  
  2133. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2134. Event: AGIExec
  2135. Privilege: agi,all
  2136. SubEvent: End
  2137. Channel: SIP/fpp-0000000b
  2138. CommandId: 1055805639
  2139. Command: GET VARIABLE cidb
  2140. ResultCode: 200
  2141. Result: Success
  2142.  
  2143.  
  2144. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'send_dtmf' is NULL
  2145. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2146. Event: AGIExec
  2147. Privilege: agi,all
  2148. SubEvent: Start
  2149. Channel: SIP/fpp-0000000b
  2150. CommandId: 897735479
  2151. Command: GET VARIABLE send_dtmf
  2152.  
  2153.  
  2154. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2155. Event: AGIExec
  2156. Privilege: agi,all
  2157. SubEvent: End
  2158. Channel: SIP/fpp-0000000b
  2159. CommandId: 897735479
  2160. Command: GET VARIABLE send_dtmf
  2161. ResultCode: 200
  2162. Result: Success
  2163.  
  2164.  
  2165. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'dtmf' is NULL
  2166. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2167. Event: AGIExec
  2168. Privilege: agi,all
  2169. SubEvent: Start
  2170. Channel: SIP/fpp-0000000b
  2171. CommandId: 293426675
  2172. Command: GET VARIABLE dtmf
  2173.  
  2174.  
  2175. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2176. Event: AGIExec
  2177. Privilege: agi,all
  2178. SubEvent: End
  2179. Channel: SIP/fpp-0000000b
  2180. CommandId: 293426675
  2181. Command: GET VARIABLE dtmf
  2182. ResultCode: 200
  2183. Result: Success
  2184.  
  2185.  
  2186. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'wava1' is NULL
  2187. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2188. Event: AGIExec
  2189. Privilege: agi,all
  2190. SubEvent: Start
  2191. Channel: SIP/fpp-0000000b
  2192. CommandId: 1607622327
  2193. Command: GET VARIABLE wava1
  2194.  
  2195.  
  2196. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2197. Event: AGIExec
  2198. Privilege: agi,all
  2199. SubEvent: End
  2200. Channel: SIP/fpp-0000000b
  2201. CommandId: 1607622327
  2202. Command: GET VARIABLE wava1
  2203. ResultCode: 200
  2204. Result: Success
  2205.  
  2206.  
  2207. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'play_wava2' is NULL
  2208. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2209. Event: AGIExec
  2210. Privilege: agi,all
  2211. SubEvent: Start
  2212. Channel: SIP/fpp-0000000b
  2213. CommandId: 1346098965
  2214. Command: GET VARIABLE play_wava2
  2215.  
  2216.  
  2217. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2218. Event: AGIExec
  2219. Privilege: agi,all
  2220. SubEvent: End
  2221. Channel: SIP/fpp-0000000b
  2222. CommandId: 1346098965
  2223. Command: GET VARIABLE play_wava2
  2224. ResultCode: 200
  2225. Result: Success
  2226.  
  2227.  
  2228. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'wava2' is NULL
  2229. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2230. Event: AGIExec
  2231. Privilege: agi,all
  2232. SubEvent: Start
  2233. Channel: SIP/fpp-0000000b
  2234. CommandId: 1053083775
  2235. Command: GET VARIABLE wava2
  2236.  
  2237.  
  2238. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2239. Event: AGIExec
  2240. Privilege: agi,all
  2241. SubEvent: End
  2242. Channel: SIP/fpp-0000000b
  2243. CommandId: 1053083775
  2244. Command: GET VARIABLE wava2
  2245. ResultCode: 200
  2246. Result: Success
  2247.  
  2248.  
  2249. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'play_wavb' is NULL
  2250. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2251. Event: AGIExec
  2252. Privilege: agi,all
  2253. SubEvent: Start
  2254. Channel: SIP/fpp-0000000b
  2255. CommandId: 161537833
  2256. Command: GET VARIABLE play_wavb
  2257.  
  2258.  
  2259. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2260. Event: AGIExec
  2261. Privilege: agi,all
  2262. SubEvent: End
  2263. Channel: SIP/fpp-0000000b
  2264. CommandId: 161537833
  2265. Command: GET VARIABLE play_wavb
  2266. ResultCode: 200
  2267. Result: Success
  2268.  
  2269.  
  2270. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'wavb' is NULL
  2271. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2272. Event: AGIExec
  2273. Privilege: agi,all
  2274. SubEvent: Start
  2275. Channel: SIP/fpp-0000000b
  2276. CommandId: 898938381
  2277. Command: GET VARIABLE wavb
  2278.  
  2279.  
  2280. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2281. Event: AGIExec
  2282. Privilege: agi,all
  2283. SubEvent: End
  2284. Channel: SIP/fpp-0000000b
  2285. CommandId: 898938381
  2286. Command: GET VARIABLE wavb
  2287. ResultCode: 200
  2288. Result: Success
  2289.  
  2290.  
  2291. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'play_wava3' is NULL
  2292. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2293. Event: AGIExec
  2294. Privilege: agi,all
  2295. SubEvent: Start
  2296. Channel: SIP/fpp-0000000b
  2297. CommandId: 603763861
  2298. Command: GET VARIABLE play_wava3
  2299.  
  2300.  
  2301. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2302. Event: AGIExec
  2303. Privilege: agi,all
  2304. SubEvent: End
  2305. Channel: SIP/fpp-0000000b
  2306. CommandId: 603763861
  2307. Command: GET VARIABLE play_wava3
  2308. ResultCode: 200
  2309. Result: Success
  2310.  
  2311.  
  2312. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'wava3' is NULL
  2313. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2314. Event: AGIExec
  2315. Privilege: agi,all
  2316. SubEvent: Start
  2317. Channel: SIP/fpp-0000000b
  2318. CommandId: 1152726757
  2319. Command: GET VARIABLE wava3
  2320.  
  2321.  
  2322. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2323. Event: AGIExec
  2324. Privilege: agi,all
  2325. SubEvent: End
  2326. Channel: SIP/fpp-0000000b
  2327. CommandId: 1152726757
  2328. Command: GET VARIABLE wava3
  2329. ResultCode: 200
  2330. Result: Success
  2331.  
  2332.  
  2333. [Jan 26 08:42:30] DEBUG[23821] pbx.c: Result of 'timeout' is NULL
  2334. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2335. Event: AGIExec
  2336. Privilege: agi,all
  2337. SubEvent: Start
  2338. Channel: SIP/fpp-0000000b
  2339. CommandId: 1955845325
  2340. Command: GET VARIABLE timeout
  2341.  
  2342.  
  2343. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2344. Event: AGIExec
  2345. Privilege: agi,all
  2346. SubEvent: End
  2347. Channel: SIP/fpp-0000000b
  2348. CommandId: 1955845325
  2349. Command: GET VARIABLE timeout
  2350. ResultCode: 200
  2351. Result: Success
  2352.  
  2353.  
  2354. [Jan 26 08:42:30] VERBOSE[23821] res_agi.c: -- <SIP/fpp-0000000b>AGI Script agi://127.0.0.1:4579/update_call_status?status=80 completed, returning 0
  2355. [Jan 26 08:42:30] VERBOSE[23821] pbx.c: -- Auto fallthrough, channel 'SIP/fpp-0000000b' status is 'UNKNOWN'
  2356. [Jan 26 08:42:30] DEBUG[23821] app_dial.c: Gosub exited with status 0
  2357. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2358. Event: VarSet
  2359. Privilege: dialplan,all
  2360. Channel: SIP/fpp-0000000b
  2361. Variable: AGISTATUS
  2362. Value: SUCCESS
  2363. Uniqueid: 1296049342.11
  2364.  
  2365.  
  2366. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: SIP answering channel: SIP/dovid-0000000a
  2367. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Setting the marker bit due to a source update
  2368. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Setting framing from config on incoming call
  2369. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  2370. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  2371. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Audio is at 5060
  2372. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  2373. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  2374. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  2375. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  2376. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c:
  2377. <--- Reliably Transmitting (NAT) to 212.7.117.61:48052 --->
  2378. SIP/2.0 200 OK
  2379. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-7b160400804d463f-1---d8754z-;received=212.7.117.61;rport=48052
  2380. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2381. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2382. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2383. CSeq: 2 INVITE
  2384. Server: Asterisk PBX 1.8.2.2
  2385. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2386. Supported: replaces, timer
  2387. Contact: <sip:10000009@208.211.92.75:5060>
  2388. Content-Type: application/sdp
  2389. Content-Length: 261
  2390.  
  2391. v=0
  2392. o=root 22860980 22860980 IN IP4 208.211.92.75
  2393. s=Asterisk PBX 1.8.2.2
  2394. c=IN IP4 208.211.92.75
  2395. t=0 0
  2396. m=audio 19710 RTP/AVP 0 101
  2397. a=rtpmap:0 PCMU/8000
  2398. a=rtpmap:101 telephone-event/8000
  2399. a=fmtp:101 0-16
  2400. a=silenceSupp:off - - - -
  2401. a=ptime:20
  2402. a=sendrecv
  2403.  
  2404. <------------>
  2405. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #110
  2406. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.7.117.61:48052
  2407. [Jan 26 08:42:30] DEBUG[23821] features.c: bridge answer set, chan answer set
  2408. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Setting the marker bit due to a source update
  2409. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Setting the marker bit due to a source update
  2410. [Jan 26 08:42:30] VERBOSE[23821] rtp_engine.c: -- Remotely bridging SIP/dovid-0000000a and SIP/fpp-0000000b
  2411. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Deferring reinvite on SIP 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' - It's audio will be redirected to IP 69.167.68.133:15760
  2412. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Sending reinvite on SIP '730bfb20211d6c7a40e584041062e145@69.167.68.130' - It's audio soon redirected to IP 192.168.1.10:53352
  2413. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  2414. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: Splitting '69.167.68.130' gives...
  2415. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  2416. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  2417. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True
  2418. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  2419. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  2420. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Audio is at 5060
  2421. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  2422. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  2423. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  2424. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  2425. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Initializing already initialized SIP dialog 730bfb20211d6c7a40e584041062e145@69.167.68.130 (presumably reinvite)
  2426. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 0 [ 46]: INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  2427. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK23ccd357
  2428. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 2 [ 48]: Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2429. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70
  2430. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 4 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2431. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 5 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2432. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 6 [ 39]: Contact: <sip:dovid@208.211.92.75:5060>
  2433. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 7 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2434. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 8 [ 16]: CSeq: 104 INVITE
  2435. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 9 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  2436. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2437. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
  2438. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  2439. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
  2440. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  2441. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  2442. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK23ccd357
  2443. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2444. Max-Forwards: 70
  2445. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2446. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2447. Contact: <sip:dovid@208.211.92.75:5060>
  2448. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2449. CSeq: 104 INVITE
  2450. User-Agent: Asterisk PBX 1.8.2.2
  2451. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2452. Supported: replaces, timer
  2453. X-asterisk-Info: SIP re-invite (External RTP bridge)
  2454. Content-Type: application/sdp
  2455. Content-Length: 263
  2456.  
  2457. v=0
  2458. o=root 1174122120 1174122122 IN IP4 192.168.1.10
  2459. s=Asterisk PBX 1.8.2.2
  2460. c=IN IP4 192.168.1.10
  2461. t=0 0
  2462. m=audio 53352 RTP/AVP 0 101
  2463. a=rtpmap:0 PCMU/8000
  2464. a=rtpmap:101 telephone-event/8000
  2465. a=fmtp:101 0-16
  2466. a=silenceSupp:off - - - -
  2467. a=ptime:20
  2468. a=sendrecv
  2469.  
  2470. ---
  2471. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #111
  2472. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  2473. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2474. Event: VarSet
  2475. Privilege: dialplan,all
  2476. Channel: SIP/dovid-0000000a
  2477. Variable: BRIDGEPEER
  2478. Value: SIP/fpp-0000000b
  2479. Uniqueid: 1296049342.10
  2480.  
  2481.  
  2482. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2483. Event: VarSet
  2484. Privilege: dialplan,all
  2485. Channel: SIP/fpp-0000000b
  2486. Variable: BRIDGEPEER
  2487. Value: SIP/dovid-0000000a
  2488. Uniqueid: 1296049342.11
  2489.  
  2490.  
  2491. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2492. Event: Newstate
  2493. Privilege: call,all
  2494. Channel: SIP/dovid-0000000a
  2495. ChannelState: 6
  2496. ChannelStateDesc: Up
  2497. CallerIDNum: dovid
  2498. CallerIDName: dovid
  2499. Uniqueid: 1296049342.10
  2500.  
  2501.  
  2502. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2503. Event: NewAccountCode
  2504. Privilege: call,all
  2505. Channel: SIP/fpp-0000000b
  2506. Uniqueid: 1296049342.11
  2507. AccountCode:
  2508. OldAccountCode:
  2509.  
  2510.  
  2511. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2512. Event: Bridge
  2513. Privilege: call,all
  2514. Bridgestate: Link
  2515. Bridgetype: core
  2516. Channel1: SIP/dovid-0000000a
  2517. Channel2: SIP/fpp-0000000b
  2518. Uniqueid1: 1296049342.10
  2519. Uniqueid2: 1296049342.11
  2520. CallerID1: dovid
  2521. CallerID2: 10000009
  2522.  
  2523.  
  2524. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2525. Event: VarSet
  2526. Privilege: dialplan,all
  2527. Channel: SIP/dovid-0000000a
  2528. Variable: BRIDGEPEER
  2529. Value: SIP/fpp-0000000b
  2530. Uniqueid: 1296049342.10
  2531.  
  2532.  
  2533. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2534. Event: VarSet
  2535. Privilege: dialplan,all
  2536. Channel: SIP/dovid-0000000a
  2537. Variable: BRIDGEPVTCALLID
  2538. Value: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2539. Uniqueid: 1296049342.10
  2540.  
  2541.  
  2542. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2543. Event: VarSet
  2544. Privilege: dialplan,all
  2545. Channel: SIP/fpp-0000000b
  2546. Variable: BRIDGEPEER
  2547. Value: SIP/dovid-0000000a
  2548. Uniqueid: 1296049342.11
  2549.  
  2550.  
  2551. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2552. Event: VarSet
  2553. Privilege: dialplan,all
  2554. Channel: SIP/fpp-0000000b
  2555. Variable: BRIDGEPVTCALLID
  2556. Value: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2557. Uniqueid: 1296049342.11
  2558.  
  2559.  
  2560. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2561. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: INVITE
  2562. [Jan 26 08:42:30] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid
  2563. [Jan 26 08:42:30] DEBUG[23716] chan_sip.c: Checking device state for peer dovid
  2564. [Jan 26 08:42:30] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use)
  2565. [Jan 26 08:42:30] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1'
  2566. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  2567. <--- SIP read from UDP:69.167.68.130:5060 --->
  2568. SIP/2.0 407 Proxy Authentication Required
  2569. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK23ccd357;rport=5060
  2570. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2571. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2572. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2573. CSeq: 104 INVITE
  2574. Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e400009c5dabad6a4835152cb164e04eec3d52c2b9"
  2575. Server: PBX_MANAGER
  2576. Content-Length: 0
  2577. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27133 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  2578.  
  2579. <------------->
  2580. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required
  2581. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK23ccd357;rport=5060
  2582. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2583. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2584. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2585. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE
  2586. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e400009c5dabad6a4835152cb164e04eec3d52c2b9"
  2587. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  2588. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0
  2589. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27133 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  2590. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) ---
  2591. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  2592. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 104
  2593. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #111
  2594. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 104: Match Found
  2595. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 407 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2596. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  2597. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  2598. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  2599. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  2600. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  2601. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  2602. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK23ccd357
  2603. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2604. Max-Forwards: 70
  2605. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2606. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2607. Contact: <sip:dovid@208.211.92.75:5060>
  2608. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2609. CSeq: 104 ACK
  2610. User-Agent: Asterisk PBX 1.8.2.2
  2611. Content-Length: 0
  2612.  
  2613.  
  2614. ---
  2615. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  2616. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE
  2617. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  2618. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  2619. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  2620. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  2621. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True
  2622. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  2623. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  2624. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Audio is at 5060
  2625. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  2626. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  2627. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP
  2628. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  2629. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  2630. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  2631. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK483d65e7
  2632. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2633. Max-Forwards: 70
  2634. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2635. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2636. Contact: <sip:dovid@208.211.92.75:5060>
  2637. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2638. CSeq: 105 INVITE
  2639. User-Agent: Asterisk PBX 1.8.2.2
  2640. Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.133:5060", nonce="4d4024e400009c5dabad6a4835152cb164e04eec3d52c2b9", response="aa056a43ecf42989b990bf98305f26eb"
  2641. Date: Wed, 26 Jan 2011 13:42:30 GMT
  2642. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2643. Supported: replaces, timer
  2644. Content-Type: application/sdp
  2645. Content-Length: 263
  2646.  
  2647. v=0
  2648. o=root 1174122120 1174122123 IN IP4 192.168.1.10
  2649. s=Asterisk PBX 1.8.2.2
  2650. c=IN IP4 192.168.1.10
  2651. t=0 0
  2652. m=audio 53352 RTP/AVP 0 101
  2653. a=rtpmap:0 PCMU/8000
  2654. a=rtpmap:101 telephone-event/8000
  2655. a=fmtp:101 0-16
  2656. a=silenceSupp:off - - - -
  2657. a=ptime:20
  2658. a=sendrecv
  2659.  
  2660. ---
  2661. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #112
  2662. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  2663. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2664. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: INVITE
  2665. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2666. Event: VarSet
  2667. Privilege: dialplan,all
  2668. Channel: SIP/dovid-0000000a
  2669. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  2670. Value: SIP 407 Proxy Authentication Required
  2671. Uniqueid: 1296049342.10
  2672.  
  2673.  
  2674. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  2675. <--- SIP read from UDP:69.167.68.130:5060 --->
  2676. SIP/2.0 100 Giving a try
  2677. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK483d65e7;rport=5060
  2678. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2679. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2680. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2681. CSeq: 105 INVITE
  2682. Server: PBX_MANAGER
  2683. Content-Length: 0
  2684. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27130 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  2685.  
  2686. <------------->
  2687. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try
  2688. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK483d65e7;rport=5060
  2689. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2690. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2691. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2692. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE
  2693. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER
  2694. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  2695. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27130 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  2696. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) ---
  2697. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  2698. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #112 - INVITE (got response)
  2699. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '730bfb20211d6c7a40e584041062e145@69.167.68.130' Request 105: Found
  2700. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 100 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2701. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2702. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: INVITE
  2703. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  2704. <--- SIP read from UDP:69.167.68.130:5060 --->
  2705. SIP/2.0 200 OK
  2706. Via: SIP/2.0/UDP 208.211.92.75:5060;rport=5060;received=208.211.92.75;branch=z9hG4bK483d65e7
  2707. Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2708. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2709. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2710. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2711. CSeq: 105 INVITE
  2712. Server: PBX_MANAGER
  2713. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  2714. Supported: replaces, timer
  2715. Contact: <sip:10000009@69.167.68.133:5060>
  2716. Content-Type: application/sdp
  2717. Content-Length: 274
  2718.  
  2719. v=0
  2720. o=root 918636038 918636039 IN IP4 69.167.68.133
  2721. s=PBX_MANAGER
  2722. c=IN IP4 69.167.68.133
  2723. t=0 0
  2724. m=audio 15760 RTP/AVP 0 101
  2725. a=rtpmap:0 PCMU/8000
  2726. a=rtpmap:101 telephone-event/8000
  2727. a=fmtp:101 0-16
  2728. a=silenceSupp:off - - - -
  2729. a=ptime:20
  2730. a=sendrecv
  2731. a=direction:active
  2732. <------------->
  2733. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  2734. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 208.211.92.75:5060;rport=5060;received=208.211.92.75;branch=z9hG4bK483d65e7
  2735. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2736. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2737. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2738. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2739. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 105 INVITE
  2740. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  2741. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  2742. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer
  2743. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: <sip:10000009@69.167.68.133:5060>
  2744. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp
  2745. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 274
  2746. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 13 [ 0]:
  2747. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  2748. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 1 [ 47]: o=root 918636038 918636039 IN IP4 69.167.68.133
  2749. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER
  2750. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.133
  2751. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  2752. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 15760 RTP/AVP 0 101
  2753. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
  2754. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  2755. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16
  2756. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - -
  2757. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 10 [ 10]: a=ptime:20
  2758. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=sendrecv
  2759. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 12 [ 18]: a=direction:active
  2760. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) ---
  2761. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  2762. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 105
  2763. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 105: Match Found
  2764. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2765. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  2766. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 918636038 918636039 IN IP4 69.167.68.133... UNSUPPORTED.
  2767. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED.
  2768. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133' gives...
  2769. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '(null)'.
  2770. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.133... OK.
  2771. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  2772. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  2773. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100
  2774. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  2775. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100
  2776. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0
  2777. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  2778. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  2779. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  2780. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
  2781. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
  2782. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  2783. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  2784. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED.
  2785. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100
  2786. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100
  2787. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  2788. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  2789. [Jan 26 08:42:30] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  2790. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.133:15760
  2791. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd055e44
  2792. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd055e44
  2793. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
  2794. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call
  2795. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call
  2796. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133:5060' gives...
  2797. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '5060'.
  2798. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  2799. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  2800. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  2801. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  2802. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  2803. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  2804. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK0e168674
  2805. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2806. Max-Forwards: 70
  2807. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2808. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2809. Contact: <sip:dovid@208.211.92.75:5060>
  2810. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2811. CSeq: 105 ACK
  2812. User-Agent: Asterisk PBX 1.8.2.2
  2813. Content-Length: 0
  2814.  
  2815.  
  2816. ---
  2817. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  2818. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2819. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: INVITE
  2820. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' changed end address to 69.167.68.133:15760 (format ulaw)
  2821. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' changed end vaddress to (null) (format ulaw)
  2822. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' changed end taddress to (null) (format ulaw)
  2823. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' was 69.167.68.133:15760/(format unknown)
  2824. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' was (null)/(format unknown)
  2825. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/fpp-0000000b' was (null)/(format unknown)
  2826. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Deferring reinvite on SIP 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' - It's audio will be redirected to IP 69.167.68.133:15760
  2827. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2828. Event: VarSet
  2829. Privilege: dialplan,all
  2830. Channel: SIP/dovid-0000000a
  2831. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  2832. Value: SIP 100 Giving a try
  2833. Uniqueid: 1296049342.10
  2834.  
  2835.  
  2836. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  2837. Event: VarSet
  2838. Privilege: dialplan,all
  2839. Channel: SIP/dovid-0000000a
  2840. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  2841. Value: SIP 200 OK
  2842. Uniqueid: 1296049342.10
  2843.  
  2844.  
  2845. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 212.7.117.61:53353
  2846. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 132 bytes
  2847. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  2848. <--- SIP read from UDP:212.7.117.61:48052 --->
  2849. ACK sip:10000009@208.211.92.75:5060 SIP/2.0
  2850. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-496e063faf585f6b-1---d8754z-;rport
  2851. Max-Forwards: 70
  2852. Contact: <sip:dovid@212.7.117.61:48052>
  2853. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2854. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2855. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2856. CSeq: 2 ACK
  2857. User-Agent: eyeBeam release 1102q stamp 51814
  2858. Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  2859. Content-Length: 0
  2860.  
  2861. <------------->
  2862. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 43]: ACK sip:10000009@208.211.92.75:5060 SIP/2.0
  2863. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-496e063faf585f6b-1---d8754z-;rport
  2864. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  2865. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  2866. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2867. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2868. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2869. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK
  2870. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  2871. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  2872. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 17]: Content-Length: 0
  2873. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (11 headers 0 lines) ---
  2874. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag as083c547c
  2875. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  2876. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110
  2877. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' of Response 2: Match Found
  2878. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Sending pending reinvite on 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.'
  2879. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Strict routing enforced for session Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2880. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:dovid@212.7.117.61:48052> for address/port to send to
  2881. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:48052' gives...
  2882. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '48052'.
  2883. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:48052
  2884. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  2885. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  2886. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  2887. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Audio is at 5060
  2888. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  2889. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  2890. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP
  2891. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  2892. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Initializing already initialized SIP dialog Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (presumably reinvite)
  2893. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 43]: INVITE sip:dovid@212.7.117.61:48052 SIP/2.0
  2894. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK099d2bdc;rport
  2895. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  2896. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 66]: From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2897. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 56]: To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2898. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 42]: Contact: <sip:10000009@208.211.92.75:5060>
  2899. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2900. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
  2901. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  2902. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2903. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer
  2904. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  2905. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
  2906. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Reliably Transmitting (NAT) to 212.7.117.61:48052:
  2907. INVITE sip:dovid@212.7.117.61:48052 SIP/2.0
  2908. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK099d2bdc;rport
  2909. Max-Forwards: 70
  2910. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2911. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2912. Contact: <sip:10000009@208.211.92.75:5060>
  2913. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2914. CSeq: 102 INVITE
  2915. User-Agent: Asterisk PBX 1.8.2.2
  2916. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  2917. Supported: replaces, timer
  2918. X-asterisk-Info: SIP re-invite (External RTP bridge)
  2919. Content-Type: application/sdp
  2920. Content-Length: 261
  2921.  
  2922. v=0
  2923. o=root 22860980 22860981 IN IP4 69.167.68.133
  2924. s=Asterisk PBX 1.8.2.2
  2925. c=IN IP4 69.167.68.133
  2926. t=0 0
  2927. m=audio 15760 RTP/AVP 0 101
  2928. a=rtpmap:0 PCMU/8000
  2929. a=rtpmap:101 telephone-event/8000
  2930. a=fmtp:101 0-16
  2931. a=silenceSupp:off - - - -
  2932. a=ptime:20
  2933. a=sendrecv
  2934.  
  2935. ---
  2936. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #113
  2937. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 212.7.117.61:48052
  2938. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2939. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  2940. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  2941. <--- SIP read from UDP:212.7.117.61:48052 --->
  2942. ACK sip:10000009@208.211.92.75:5060 SIP/2.0
  2943. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-496e063faf585f6b-1---d8754z-;rport
  2944. Max-Forwards: 70
  2945. Contact: <sip:dovid@212.7.117.61:48052>
  2946. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2947. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2948. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2949. CSeq: 2 ACK
  2950. User-Agent: eyeBeam release 1102q stamp 51814
  2951. Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  2952. Content-Length: 0
  2953.  
  2954. <------------->
  2955. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 43]: ACK sip:10000009@208.211.92.75:5060 SIP/2.0
  2956. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-496e063faf585f6b-1---d8754z-;rport
  2957. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  2958. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  2959. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  2960. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  2961. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  2962. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK
  2963. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  2964. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@mypbx.mydomain.com",response="69035707b61056b23c73c5d287ead7eb",algorithm=MD5
  2965. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 17]: Content-Length: 0
  2966. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (11 headers 0 lines) ---
  2967. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag as083c547c
  2968. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  2969. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  2970. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  2971. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  2972. [Jan 26 08:42:30] DEBUG[23821] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 212.7.117.61:53352
  2973. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/dovid-0000000a' changed end address to 212.7.117.61:53352 (format unknown)
  2974. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/dovid-0000000a' was 192.168.1.10:53352/(format unknown)
  2975. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Sending reinvite on SIP '730bfb20211d6c7a40e584041062e145@69.167.68.130' - It's audio soon redirected to IP 212.7.117.61:53352
  2976. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  2977. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: Splitting '69.167.68.130' gives...
  2978. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  2979. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  2980. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  2981. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  2982. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  2983. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Audio is at 5060
  2984. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  2985. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  2986. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  2987. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  2988. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Initializing already initialized SIP dialog 730bfb20211d6c7a40e584041062e145@69.167.68.130 (presumably reinvite)
  2989. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 0 [ 46]: INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  2990. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK667ec2f5
  2991. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 2 [ 48]: Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  2992. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70
  2993. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 4 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  2994. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 5 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  2995. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 6 [ 39]: Contact: <sip:dovid@208.211.92.75:5060>
  2996. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 7 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  2997. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 8 [ 16]: CSeq: 106 INVITE
  2998. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 9 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  2999. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3000. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
  3001. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  3002. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
  3003. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  3004. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  3005. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK667ec2f5
  3006. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3007. Max-Forwards: 70
  3008. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3009. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3010. Contact: <sip:dovid@208.211.92.75:5060>
  3011. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3012. CSeq: 106 INVITE
  3013. User-Agent: Asterisk PBX 1.8.2.2
  3014. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3015. Supported: replaces, timer
  3016. X-asterisk-Info: SIP re-invite (External RTP bridge)
  3017. Content-Type: application/sdp
  3018. Content-Length: 263
  3019.  
  3020. v=0
  3021. o=root 1174122120 1174122124 IN IP4 212.7.117.61
  3022. s=Asterisk PBX 1.8.2.2
  3023. c=IN IP4 212.7.117.61
  3024. t=0 0
  3025. m=audio 53352 RTP/AVP 0 101
  3026. a=rtpmap:0 PCMU/8000
  3027. a=rtpmap:101 telephone-event/8000
  3028. a=fmtp:101 0-16
  3029. a=silenceSupp:off - - - -
  3030. a=ptime:20
  3031. a=sendrecv
  3032.  
  3033. ---
  3034. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #114
  3035. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  3036. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3037. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3038. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3039. <--- SIP read from UDP:69.167.68.130:5060 --->
  3040. SIP/2.0 407 Proxy Authentication Required
  3041. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK667ec2f5
  3042. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3043. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3044. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3045. CSeq: 106 INVITE
  3046. Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e400009c68b33ebbf47ec744b011cce366bcd377a3"
  3047. Server: PBX_MANAGER
  3048. Content-Length: 0
  3049. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3050.  
  3051. <------------->
  3052. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required
  3053. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK667ec2f5
  3054. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3055. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3056. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3057. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 106 INVITE
  3058. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e400009c68b33ebbf47ec744b011cce366bcd377a3"
  3059. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  3060. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0
  3061. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3062. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) ---
  3063. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3064. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 106
  3065. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #114
  3066. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 106: Match Found
  3067. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 407 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3068. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3069. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3070. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3071. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3072. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  3073. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  3074. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK667ec2f5
  3075. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3076. Max-Forwards: 70
  3077. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3078. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3079. Contact: <sip:dovid@208.211.92.75:5060>
  3080. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3081. CSeq: 106 ACK
  3082. User-Agent: Asterisk PBX 1.8.2.2
  3083. Content-Length: 0
  3084.  
  3085.  
  3086. ---
  3087. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  3088. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE
  3089. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3090. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3091. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3092. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3093. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  3094. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  3095. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  3096. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Audio is at 5060
  3097. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  3098. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  3099. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP
  3100. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  3101. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  3102. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  3103. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f21742b
  3104. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3105. Max-Forwards: 70
  3106. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3107. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3108. Contact: <sip:dovid@208.211.92.75:5060>
  3109. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3110. CSeq: 107 INVITE
  3111. User-Agent: Asterisk PBX 1.8.2.2
  3112. Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.133:5060", nonce="4d4024e400009c68b33ebbf47ec744b011cce366bcd377a3", response="36f5d572e969ffa830d5b9479cb05bad"
  3113. Date: Wed, 26 Jan 2011 13:42:30 GMT
  3114. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3115. Supported: replaces, timer
  3116. Content-Type: application/sdp
  3117. Content-Length: 263
  3118.  
  3119. v=0
  3120. o=root 1174122120 1174122125 IN IP4 212.7.117.61
  3121. s=Asterisk PBX 1.8.2.2
  3122. c=IN IP4 212.7.117.61
  3123. t=0 0
  3124. m=audio 53352 RTP/AVP 0 101
  3125. a=rtpmap:0 PCMU/8000
  3126. a=rtpmap:101 telephone-event/8000
  3127. a=fmtp:101 0-16
  3128. a=silenceSupp:off - - - -
  3129. a=ptime:20
  3130. a=sendrecv
  3131.  
  3132. ---
  3133. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #115
  3134. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  3135. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3136. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3137. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3138. Event: VarSet
  3139. Privilege: dialplan,all
  3140. Channel: SIP/dovid-0000000a
  3141. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3142. Value: SIP 407 Proxy Authentication Required
  3143. Uniqueid: 1296049342.10
  3144.  
  3145.  
  3146. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3147. <--- SIP read from UDP:69.167.68.130:5060 --->
  3148. SIP/2.0 100 Giving a try
  3149. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f21742b
  3150. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3151. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3152. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3153. CSeq: 107 INVITE
  3154. Server: PBX_MANAGER
  3155. Content-Length: 0
  3156. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3157.  
  3158. <------------->
  3159. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try
  3160. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f21742b
  3161. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3162. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3163. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3164. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 107 INVITE
  3165. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER
  3166. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  3167. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3168. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) ---
  3169. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3170. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #115 - INVITE (got response)
  3171. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '730bfb20211d6c7a40e584041062e145@69.167.68.130' Request 107: Found
  3172. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 100 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3173. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3174. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3175. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3176. Event: VarSet
  3177. Privilege: dialplan,all
  3178. Channel: SIP/dovid-0000000a
  3179. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3180. Value: SIP 100 Giving a try
  3181. Uniqueid: 1296049342.10
  3182.  
  3183.  
  3184. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3185. <--- SIP read from UDP:69.167.68.130:5060 --->
  3186. SIP/2.0 200 OK
  3187. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f21742b
  3188. Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3189. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3190. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3191. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3192. CSeq: 107 INVITE
  3193. Server: PBX_MANAGER
  3194. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  3195. Supported: replaces, timer
  3196. Contact: <sip:10000009@69.167.68.133:5060>
  3197. Content-Type: application/sdp
  3198. Content-Length: 274
  3199.  
  3200. v=0
  3201. o=root 918636038 918636040 IN IP4 69.167.68.133
  3202. s=PBX_MANAGER
  3203. c=IN IP4 69.167.68.133
  3204. t=0 0
  3205. m=audio 15760 RTP/AVP 0 101
  3206. a=rtpmap:0 PCMU/8000
  3207. a=rtpmap:101 telephone-event/8000
  3208. a=fmtp:101 0-16
  3209. a=silenceSupp:off - - - -
  3210. a=ptime:20
  3211. a=sendrecv
  3212. a=direction:active
  3213. <------------->
  3214. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  3215. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f21742b
  3216. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3217. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3218. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3219. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3220. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 107 INVITE
  3221. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  3222. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  3223. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer
  3224. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: <sip:10000009@69.167.68.133:5060>
  3225. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp
  3226. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 274
  3227. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 13 [ 0]:
  3228. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  3229. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 1 [ 47]: o=root 918636038 918636040 IN IP4 69.167.68.133
  3230. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER
  3231. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.133
  3232. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  3233. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 15760 RTP/AVP 0 101
  3234. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
  3235. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  3236. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16
  3237. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - -
  3238. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 10 [ 10]: a=ptime:20
  3239. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=sendrecv
  3240. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 12 [ 18]: a=direction:active
  3241. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) ---
  3242. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3243. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 107
  3244. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 107: Match Found
  3245. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3246. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  3247. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 918636038 918636040 IN IP4 69.167.68.133... UNSUPPORTED.
  3248. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED.
  3249. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133' gives...
  3250. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '(null)'.
  3251. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.133... OK.
  3252. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  3253. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  3254. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100
  3255. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  3256. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100
  3257. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0
  3258. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  3259. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  3260. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  3261. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
  3262. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
  3263. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  3264. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  3265. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED.
  3266. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100
  3267. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100
  3268. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  3269. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  3270. [Jan 26 08:42:30] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  3271. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.133:15760
  3272. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd055e44
  3273. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd055e44
  3274. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
  3275. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call
  3276. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call
  3277. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133:5060' gives...
  3278. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '5060'.
  3279. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3280. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3281. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3282. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3283. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  3284. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  3285. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4efb1091
  3286. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3287. Max-Forwards: 70
  3288. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3289. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3290. Contact: <sip:dovid@208.211.92.75:5060>
  3291. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3292. CSeq: 107 ACK
  3293. User-Agent: Asterisk PBX 1.8.2.2
  3294. Content-Length: 0
  3295.  
  3296.  
  3297. ---
  3298. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  3299. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3300. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3301. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3302. Event: VarSet
  3303. Privilege: dialplan,all
  3304. Channel: SIP/dovid-0000000a
  3305. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3306. Value: SIP 200 OK
  3307. Uniqueid: 1296049342.10
  3308.  
  3309.  
  3310. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3311. <--- SIP read from UDP:212.7.117.61:48052 --->
  3312. SIP/2.0 200 OK
  3313. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK099d2bdc;rport=5060
  3314. Contact: <sip:dovid@212.7.117.61:48052>
  3315. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  3316. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  3317. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3318. CSeq: 102 INVITE
  3319. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  3320. Content-Type: application/sdp
  3321. User-Agent: eyeBeam release 1102q stamp 51814
  3322. Content-Length: 184
  3323.  
  3324. v=0
  3325. o=- 9 3 IN IP4 192.168.1.10
  3326. s=CounterPath eyeBeam 1.5
  3327. c=IN IP4 192.168.1.10
  3328. t=0 0
  3329. m=audio 53352 RTP/AVP 0 101
  3330. a=fmtp:101 0-15
  3331. a=rtpmap:101 telephone-event/8000
  3332. a=sendrecv
  3333. <------------->
  3334. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  3335. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK099d2bdc;rport=5060
  3336. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  3337. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 56]: To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  3338. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 66]: From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  3339. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3340. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE
  3341. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  3342. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp
  3343. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  3344. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 19]: Content-Length: 184
  3345. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 11 [ 0]:
  3346. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  3347. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 9 3 IN IP4 192.168.1.10
  3348. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5
  3349. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.10
  3350. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  3351. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 53352 RTP/AVP 0 101
  3352. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15
  3353. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  3354. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 8 [ 10]: a=sendrecv
  3355. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (11 headers 9 lines) ---
  3356. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking To) --From tag as083c547c --To-tag a23db027
  3357. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 102
  3358. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #113
  3359. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' of Request 102: Match Found
  3360. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3361. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  3362. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP o=- 9 3 IN IP4 192.168.1.10... UNSUPPORTED.
  3363. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED.
  3364. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '192.168.1.10' gives...
  3365. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '(null)'.
  3366. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.10... OK.
  3367. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  3368. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  3369. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100
  3370. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  3371. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100
  3372. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
  3373. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  3374. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  3375. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  3376. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100
  3377. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100
  3378. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  3379. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  3380. [Jan 26 08:42:30] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  3381. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 192.168.1.10:53352
  3382. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd050dac
  3383. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd050dac
  3384. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
  3385. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call
  3386. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Updating call counter for incoming call
  3387. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Strict routing enforced for session Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3388. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:dovid@212.7.117.61:48052> for address/port to send to
  3389. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:48052' gives...
  3390. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '48052'.
  3391. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:48052
  3392. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (NAT) to 212.7.117.61:48052:
  3393. ACK sip:dovid@212.7.117.61:48052 SIP/2.0
  3394. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK1b1d2f52;rport
  3395. Max-Forwards: 70
  3396. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  3397. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  3398. Contact: <sip:10000009@208.211.92.75:5060>
  3399. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3400. CSeq: 102 ACK
  3401. User-Agent: Asterisk PBX 1.8.2.2
  3402. Content-Length: 0
  3403.  
  3404.  
  3405. ---
  3406. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:dov' onto UDP socket destined for 212.7.117.61:48052
  3407. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3408. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3409. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/dovid-0000000a' changed end address to 192.168.1.10:53352 (format ulaw)
  3410. [Jan 26 08:42:30] DEBUG[23821] rtp_engine.c: Oooh, 'SIP/dovid-0000000a' was 212.7.117.61:53352/(format unknown)
  3411. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Sending reinvite on SIP '730bfb20211d6c7a40e584041062e145@69.167.68.130' - It's audio soon redirected to IP 192.168.1.10:53352
  3412. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3413. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: Splitting '69.167.68.130' gives...
  3414. [Jan 26 08:42:30] DEBUG[23821] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3415. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3416. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  3417. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  3418. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  3419. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Audio is at 5060
  3420. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  3421. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  3422. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  3423. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  3424. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Initializing already initialized SIP dialog 730bfb20211d6c7a40e584041062e145@69.167.68.130 (presumably reinvite)
  3425. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 0 [ 46]: INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  3426. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6461a0d3
  3427. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 2 [ 48]: Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3428. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70
  3429. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 4 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3430. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 5 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3431. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 6 [ 39]: Contact: <sip:dovid@208.211.92.75:5060>
  3432. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 7 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3433. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 8 [ 16]: CSeq: 108 INVITE
  3434. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 9 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  3435. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3436. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
  3437. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  3438. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
  3439. [Jan 26 08:42:30] VERBOSE[23821] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  3440. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  3441. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6461a0d3
  3442. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3443. Max-Forwards: 70
  3444. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3445. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3446. Contact: <sip:dovid@208.211.92.75:5060>
  3447. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3448. CSeq: 108 INVITE
  3449. User-Agent: Asterisk PBX 1.8.2.2
  3450. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3451. Supported: replaces, timer
  3452. X-asterisk-Info: SIP re-invite (External RTP bridge)
  3453. Content-Type: application/sdp
  3454. Content-Length: 263
  3455.  
  3456. v=0
  3457. o=root 1174122120 1174122126 IN IP4 192.168.1.10
  3458. s=Asterisk PBX 1.8.2.2
  3459. c=IN IP4 192.168.1.10
  3460. t=0 0
  3461. m=audio 53352 RTP/AVP 0 101
  3462. a=rtpmap:0 PCMU/8000
  3463. a=rtpmap:101 telephone-event/8000
  3464. a=fmtp:101 0-16
  3465. a=silenceSupp:off - - - -
  3466. a=ptime:20
  3467. a=sendrecv
  3468.  
  3469. ---
  3470. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #116
  3471. [Jan 26 08:42:30] DEBUG[23821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  3472. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3473. Event: VarSet
  3474. Privilege: dialplan,all
  3475. Channel: SIP/dovid-0000000a
  3476. Variable: ~HASH~SIP_CAUSE~SIP/dovid-0000000a~
  3477. Value: SIP 200 OK
  3478. Uniqueid: 1296049342.10
  3479.  
  3480.  
  3481. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3482. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3483. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3484. <--- SIP read from UDP:69.167.68.130:5060 --->
  3485. SIP/2.0 407 Proxy Authentication Required
  3486. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6461a0d3;rport=5060
  3487. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3488. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3489. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3490. CSeq: 108 INVITE
  3491. Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e500009c730786ec4b4bccd3261e22e76bc6849e1d"
  3492. Server: PBX_MANAGER
  3493. Content-Length: 0
  3494. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27127 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3495.  
  3496. <------------->
  3497. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required
  3498. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6461a0d3;rport=5060
  3499. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3500. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3501. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3502. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 108 INVITE
  3503. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4024e500009c730786ec4b4bccd3261e22e76bc6849e1d"
  3504. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  3505. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0
  3506. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27127 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3507. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) ---
  3508. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3509. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 108
  3510. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #116
  3511. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 108: Match Found
  3512. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 407 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3513. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3514. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3515. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3516. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3517. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  3518. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  3519. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK6461a0d3
  3520. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3521. Max-Forwards: 70
  3522. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3523. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3524. Contact: <sip:dovid@208.211.92.75:5060>
  3525. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3526. CSeq: 108 ACK
  3527. User-Agent: Asterisk PBX 1.8.2.2
  3528. Content-Length: 0
  3529.  
  3530.  
  3531. ---
  3532. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  3533. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE
  3534. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3535. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3536. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3537. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3538. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  3539. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
  3540. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw)
  3541. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Audio is at 5060
  3542. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  3543. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  3544. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP
  3545. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  3546. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060:
  3547. INVITE sip:10000009@69.167.68.133:5060 SIP/2.0
  3548. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK047e261e
  3549. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3550. Max-Forwards: 70
  3551. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3552. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3553. Contact: <sip:dovid@208.211.92.75:5060>
  3554. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3555. CSeq: 109 INVITE
  3556. User-Agent: Asterisk PBX 1.8.2.2
  3557. Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.133:5060", nonce="4d4024e500009c730786ec4b4bccd3261e22e76bc6849e1d", response="b8759655c856194a69e08f85ff8816e5"
  3558. Date: Wed, 26 Jan 2011 13:42:30 GMT
  3559. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3560. Supported: replaces, timer
  3561. Content-Type: application/sdp
  3562. Content-Length: 263
  3563.  
  3564. v=0
  3565. o=root 1174122120 1174122127 IN IP4 192.168.1.10
  3566. s=Asterisk PBX 1.8.2.2
  3567. c=IN IP4 192.168.1.10
  3568. t=0 0
  3569. m=audio 53352 RTP/AVP 0 101
  3570. a=rtpmap:0 PCMU/8000
  3571. a=rtpmap:101 telephone-event/8000
  3572. a=fmtp:101 0-16
  3573. a=silenceSupp:off - - - -
  3574. a=ptime:20
  3575. a=sendrecv
  3576.  
  3577. ---
  3578. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #117
  3579. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060
  3580. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3581. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3582. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3583. Event: VarSet
  3584. Privilege: dialplan,all
  3585. Channel: SIP/dovid-0000000a
  3586. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3587. Value: SIP 407 Proxy Authentication Required
  3588. Uniqueid: 1296049342.10
  3589.  
  3590.  
  3591. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3592. <--- SIP read from UDP:69.167.68.130:5060 --->
  3593. SIP/2.0 100 Giving a try
  3594. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK047e261e;rport=5060
  3595. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3596. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3597. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3598. CSeq: 109 INVITE
  3599. Server: PBX_MANAGER
  3600. Content-Length: 0
  3601. Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3602.  
  3603. <------------->
  3604. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try
  3605. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK047e261e;rport=5060
  3606. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3607. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3608. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3609. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 109 INVITE
  3610. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER
  3611. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  3612. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.133:5060 out_uri=sip:10000009@69.167.68.133:5060 via_cnt==1"
  3613. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) ---
  3614. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3615. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #117 - INVITE (got response)
  3616. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '730bfb20211d6c7a40e584041062e145@69.167.68.130' Request 109: Found
  3617. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 100 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3618. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3619. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3620. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c:
  3621. <--- SIP read from UDP:69.167.68.130:5060 --->
  3622. SIP/2.0 200 OK
  3623. Via: SIP/2.0/UDP 208.211.92.75:5060;rport=5060;received=208.211.92.75;branch=z9hG4bK047e261e
  3624. Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3625. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3626. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3627. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3628. CSeq: 109 INVITE
  3629. Server: PBX_MANAGER
  3630. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  3631. Supported: replaces, timer
  3632. Contact: <sip:10000009@69.167.68.133:5060>
  3633. Content-Type: application/sdp
  3634. Content-Length: 274
  3635.  
  3636. v=0
  3637. o=root 918636038 918636041 IN IP4 69.167.68.133
  3638. s=PBX_MANAGER
  3639. c=IN IP4 69.167.68.133
  3640. t=0 0
  3641. m=audio 15760 RTP/AVP 0 101
  3642. a=rtpmap:0 PCMU/8000
  3643. a=rtpmap:101 telephone-event/8000
  3644. a=fmtp:101 0-16
  3645. a=silenceSupp:off - - - -
  3646. a=ptime:20
  3647. a=sendrecv
  3648. a=direction:active
  3649. <------------->
  3650. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  3651. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 208.211.92.75:5060;rport=5060;received=208.211.92.75;branch=z9hG4bK047e261e
  3652. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3653. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3654. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3655. Event: VarSet
  3656. Privilege: dialplan,all
  3657. Channel: SIP/dovid-0000000a
  3658. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3659. Value: SIP 100 Giving a try
  3660. Uniqueid: 1296049342.10
  3661.  
  3662.  
  3663. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3664. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3665. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 109 INVITE
  3666. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER
  3667. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  3668. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer
  3669. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: <sip:10000009@69.167.68.133:5060>
  3670. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp
  3671. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 274
  3672. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Header 13 [ 0]:
  3673. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  3674. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 1 [ 47]: o=root 918636038 918636041 IN IP4 69.167.68.133
  3675. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER
  3676. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.133
  3677. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  3678. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 15760 RTP/AVP 0 101
  3679. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
  3680. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  3681. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16
  3682. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - -
  3683. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 10 [ 10]: a=ptime:20
  3684. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=sendrecv
  3685. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Body 12 [ 18]: a=direction:active
  3686. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) ---
  3687. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking To) --From tag as72ac6b1e --To-tag as1f849b69
  3688. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Acked pending invite 109
  3689. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Stopping retransmission on '730bfb20211d6c7a40e584041062e145@69.167.68.130' of Request 109: Match Found
  3690. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3691. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  3692. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 918636038 918636041 IN IP4 69.167.68.133... UNSUPPORTED.
  3693. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED.
  3694. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133' gives...
  3695. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '(null)'.
  3696. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.133... OK.
  3697. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  3698. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 0
  3699. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100
  3700. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found RTP audio format 101
  3701. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100
  3702. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0
  3703. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  3704. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101
  3705. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  3706. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
  3707. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
  3708. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  3709. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
  3710. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED.
  3711. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100
  3712. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100
  3713. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  3714. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  3715. [Jan 26 08:42:30] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  3716. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.133:15760
  3717. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd055e44
  3718. [Jan 26 08:42:30] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd055e44
  3719. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
  3720. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call
  3721. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call
  3722. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.133:5060' gives...
  3723. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.133' and port '5060'.
  3724. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:69.167.68.130;lr=on;ftag=as72ac6b1e> for address/port to send to
  3725. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3726. [Jan 26 08:42:30] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3727. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060
  3728. [Jan 26 08:42:30] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060:
  3729. ACK sip:10000009@69.167.68.133:5060 SIP/2.0
  3730. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK61e7598e
  3731. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3732. Max-Forwards: 70
  3733. From: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3734. To: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3735. Contact: <sip:dovid@208.211.92.75:5060>
  3736. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3737. CSeq: 109 ACK
  3738. User-Agent: Asterisk PBX 1.8.2.2
  3739. Content-Length: 0
  3740.  
  3741.  
  3742. ---
  3743. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060
  3744. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3745. [Jan 26 08:42:30] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3746. [Jan 26 08:42:30] DEBUG[23750] manager.c: Examining event:
  3747. Event: VarSet
  3748. Privilege: dialplan,all
  3749. Channel: SIP/dovid-0000000a
  3750. Variable: ~HASH~SIP_CAUSE~SIP/fpp-0000000b~
  3751. Value: SIP 200 OK
  3752. Uniqueid: 1296049342.10
  3753.  
  3754.  
  3755. [Jan 26 08:42:31] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3756. [Jan 26 08:42:31] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3757. [Jan 26 08:42:32] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3758. [Jan 26 08:42:32] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3759. [Jan 26 08:42:33] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3760. [Jan 26 08:42:33] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3761. [Jan 26 08:42:34] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3762. [Jan 26 08:42:34] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3763. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: INVITE
  3764. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3765. [Jan 26 08:42:35] VERBOSE[23737] chan_sip.c:
  3766. <--- SIP read from UDP:69.167.68.130:5060 --->
  3767. BYE sip:dovid@208.211.92.75:5060 SIP/2.0
  3768. Record-Route: <sip:69.167.68.130;lr=on;ftag=as1f849b69>
  3769. Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbecb.8c92cf55.0
  3770. Via: SIP/2.0/UDP 69.167.68.133:5060;received=69.167.68.133;branch=z9hG4bK4a48b7fd;rport=5060
  3771. Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3772. Max-Forwards: 69
  3773. From: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3774. To: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3775. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3776. CSeq: 102 BYE
  3777. User-Agent: PBX_MANAGER
  3778. X-Asterisk-HangupCause: Normal Clearing
  3779. X-Asterisk-HangupCauseCode: 16
  3780. Content-Length: 0
  3781. X-Enswitch-RURI: sip:dovid@208.211.92.75:5060
  3782. X-Enswitch-Source: 69.167.68.133:5060
  3783.  
  3784. <------------->
  3785. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 0 [ 40]: BYE sip:dovid@208.211.92.75:5060 SIP/2.0
  3786. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 1 [ 55]: Record-Route: <sip:69.167.68.130;lr=on;ftag=as1f849b69>
  3787. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 2 [ 60]: Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbecb.8c92cf55.0
  3788. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 3 [ 92]: Via: SIP/2.0/UDP 69.167.68.133:5060;received=69.167.68.133;branch=z9hG4bK4a48b7fd;rport=5060
  3789. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 4 [ 48]: Route: <sip:69.167.68.130;lr=on;ftag=as72ac6b1e>
  3790. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 5 [ 16]: Max-Forwards: 69
  3791. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 6 [ 49]: From: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3792. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 7 [ 52]: To: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3793. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 8 [ 55]: Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3794. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 9 [ 13]: CSeq: 102 BYE
  3795. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 10 [ 23]: User-Agent: PBX_MANAGER
  3796. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 11 [ 39]: X-Asterisk-HangupCause: Normal Clearing
  3797. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 12 [ 30]: X-Asterisk-HangupCauseCode: 16
  3798. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 13 [ 17]: Content-Length: 0
  3799. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 14 [ 45]: X-Enswitch-RURI: sip:dovid@208.211.92.75:5060
  3800. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Header 15 [ 37]: X-Enswitch-Source: 69.167.68.133:5060
  3801. [Jan 26 08:42:35] VERBOSE[23737] chan_sip.c: --- (16 headers 0 lines) ---
  3802. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: = Looking for Call ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130 (Checking From) --From tag as1f849b69 --To-tag as72ac6b1e
  3803. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
  3804. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Initializing initreq for method BYE - callid 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3805. [Jan 26 08:42:35] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives...
  3806. [Jan 26 08:42:35] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'.
  3807. [Jan 26 08:42:35] VERBOSE[23737] chan_sip.c: Sending to 69.167.68.130:5060 (no NAT)
  3808. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Setting SIP_ALREADYGONE on dialog 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3809. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3810. Event: VarSet
  3811. Privilege: dialplan,all
  3812. Channel: SIP/fpp-0000000b
  3813. Variable: RTPAUDIOQOS
  3814. Value: ssrc=529697820;themssrc=209841337;lp=0;rxjitter=0.002752;rxcount=248;txjitter=0.000000;txcount=141;rlp=0;rtt=0.000000
  3815. Uniqueid: 1296049342.11
  3816.  
  3817.  
  3818. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3819. Event: VarSet
  3820. Privilege: dialplan,all
  3821. Channel: SIP/dovid-0000000a
  3822. Variable: RTPAUDIOQOSBRIDGED
  3823. Value: ssrc=529697820;themssrc=209841337;lp=0;rxjitter=0.002752;rxcount=248;txjitter=0.000000;txcount=141;rlp=0;rtt=0.000000
  3824. Uniqueid: 1296049342.10
  3825.  
  3826.  
  3827. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3828. Event: VarSet
  3829. Privilege: dialplan,all
  3830. Channel: SIP/fpp-0000000b
  3831. Variable: RTPAUDIOQOSJITTER
  3832. Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
  3833. Uniqueid: 1296049342.11
  3834.  
  3835.  
  3836. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3837. Event: VarSet
  3838. Privilege: dialplan,all
  3839. Channel: SIP/dovid-0000000a
  3840. Variable: RTPAUDIOQOSJITTERBRIDGED
  3841. Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
  3842. Uniqueid: 1296049342.10
  3843.  
  3844.  
  3845. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3846. Event: VarSet
  3847. Privilege: dialplan,all
  3848. Channel: SIP/fpp-0000000b
  3849. Variable: RTPAUDIOQOSLOSS
  3850. Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
  3851. Uniqueid: 1296049342.11
  3852.  
  3853.  
  3854. [Jan 26 08:42:35] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  3855. [Jan 26 08:42:35] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' in 32000 ms (Method: BYE)
  3856. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Received bye, issuing owner hangup
  3857. [Jan 26 08:42:35] VERBOSE[23737] chan_sip.c:
  3858. <--- Transmitting (no NAT) to 69.167.68.130:5060 --->
  3859. SIP/2.0 200 OK
  3860. Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbecb.8c92cf55.0;received=69.167.68.130
  3861. Via: SIP/2.0/UDP 69.167.68.133:5060;received=69.167.68.133;branch=z9hG4bK4a48b7fd;rport=5060
  3862. Record-Route: <sip:69.167.68.130;lr=on;ftag=as1f849b69>
  3863. From: <sip:10000009@69.167.68.130>;tag=as1f849b69
  3864. To: "dovid" <sip:dovid@69.167.68.130>;tag=as72ac6b1e
  3865. Call-ID: 730bfb20211d6c7a40e584041062e145@69.167.68.130
  3866. CSeq: 102 BYE
  3867. Server: Asterisk PBX 1.8.2.2
  3868. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  3869. Supported: replaces, timer
  3870. Content-Length: 0
  3871.  
  3872.  
  3873. <------------>
  3874. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 69.167.68.130:5060
  3875. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '730bfb20211d6c7a40e584041062e145@69.167.68.130' Method: BYE
  3876. [Jan 26 08:42:35] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' Method: ACK
  3877. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3878. Event: VarSet
  3879. Privilege: dialplan,all
  3880. Channel: SIP/dovid-0000000a
  3881. Variable: RTPAUDIOQOSLOSSBRIDGED
  3882. Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
  3883. Uniqueid: 1296049342.10
  3884.  
  3885.  
  3886. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3887. Event: VarSet
  3888. Privilege: dialplan,all
  3889. Channel: SIP/fpp-0000000b
  3890. Variable: RTPAUDIOQOSRTT
  3891. Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
  3892. Uniqueid: 1296049342.11
  3893.  
  3894.  
  3895. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3896. Event: VarSet
  3897. Privilege: dialplan,all
  3898. Channel: SIP/dovid-0000000a
  3899. Variable: RTPAUDIOQOSRTTBRIDGED
  3900. Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
  3901. Uniqueid: 1296049342.10
  3902.  
  3903.  
  3904. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3905. Event: VarSet
  3906. Privilege: dialplan,all
  3907. Channel: SIP/dovid-0000000a
  3908. Variable: RTPAUDIOQOS
  3909. Value: ssrc=1901277685;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
  3910. Uniqueid: 1296049342.10
  3911.  
  3912.  
  3913. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3914. Event: VarSet
  3915. Privilege: dialplan,all
  3916. Channel: SIP/fpp-0000000b
  3917. Variable: RTPAUDIOQOSBRIDGED
  3918. Value: ssrc=1901277685;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
  3919. Uniqueid: 1296049342.11
  3920.  
  3921.  
  3922. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3923. Event: VarSet
  3924. Privilege: dialplan,all
  3925. Channel: SIP/dovid-0000000a
  3926. Variable: RTPAUDIOQOSJITTER
  3927. Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
  3928. Uniqueid: 1296049342.10
  3929.  
  3930.  
  3931. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3932. Event: VarSet
  3933. Privilege: dialplan,all
  3934. Channel: SIP/fpp-0000000b
  3935. Variable: RTPAUDIOQOSJITTERBRIDGED
  3936. Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
  3937. Uniqueid: 1296049342.11
  3938.  
  3939.  
  3940. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3941. Event: VarSet
  3942. Privilege: dialplan,all
  3943. Channel: SIP/dovid-0000000a
  3944. Variable: RTPAUDIOQOSLOSS
  3945. Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
  3946. Uniqueid: 1296049342.10
  3947.  
  3948.  
  3949. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3950. Event: VarSet
  3951. Privilege: dialplan,all
  3952. Channel: SIP/fpp-0000000b
  3953. Variable: RTPAUDIOQOSLOSSBRIDGED
  3954. Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
  3955. Uniqueid: 1296049342.11
  3956.  
  3957.  
  3958. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3959. Event: VarSet
  3960. Privilege: dialplan,all
  3961. Channel: SIP/dovid-0000000a
  3962. Variable: RTPAUDIOQOSRTT
  3963. Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
  3964. Uniqueid: 1296049342.10
  3965.  
  3966.  
  3967. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  3968. Event: VarSet
  3969. Privilege: dialplan,all
  3970. Channel: SIP/fpp-0000000b
  3971. Variable: RTPAUDIOQOSRTTBRIDGED
  3972. Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
  3973. Uniqueid: 1296049342.11
  3974.  
  3975.  
  3976. [Jan 26 08:42:35] DEBUG[23821] rtp_engine.c: Oooh, got a hangup
  3977. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Sending reinvite on SIP 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' - It's audio soon redirected to IP 208.211.92.75:5060
  3978. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Strict routing enforced for session Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3979. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: set_destination: Parsing <sip:dovid@212.7.117.61:48052> for address/port to send to
  3980. [Jan 26 08:42:35] DEBUG[23821] netsock2.c: Splitting '212.7.117.61:48052' gives...
  3981. [Jan 26 08:42:35] DEBUG[23821] netsock2.c: ...host '212.7.117.61' and port '48052'.
  3982. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: set_destination: set destination to 212.7.117.61:48052
  3983. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
  3984. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  3985. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: Audio is at 5060
  3986. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  3987. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  3988. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: -- Done with adding codecs to SDP
  3989. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
  3990. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Initializing already initialized SIP dialog Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (presumably reinvite)
  3991. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 0 [ 43]: INVITE sip:dovid@212.7.117.61:48052 SIP/2.0
  3992. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4103729c;rport
  3993. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  3994. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 3 [ 66]: From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  3995. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 4 [ 56]: To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  3996. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 5 [ 42]: Contact: <sip:10000009@208.211.92.75:5060>
  3997. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  3998. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE
  3999. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2
  4000. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  4001. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer
  4002. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  4003. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
  4004. [Jan 26 08:42:35] VERBOSE[23821] chan_sip.c: Reliably Transmitting (NAT) to 212.7.117.61:48052:
  4005. INVITE sip:dovid@212.7.117.61:48052 SIP/2.0
  4006. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4103729c;rport
  4007. Max-Forwards: 70
  4008. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4009. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4010. Contact: <sip:10000009@208.211.92.75:5060>
  4011. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4012. CSeq: 103 INVITE
  4013. User-Agent: Asterisk PBX 1.8.2.2
  4014. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  4015. Supported: replaces, timer
  4016. X-asterisk-Info: SIP re-invite (External RTP bridge)
  4017. Content-Type: application/sdp
  4018. Content-Length: 261
  4019.  
  4020. v=0
  4021. o=root 22860980 22860982 IN IP4 208.211.92.75
  4022. s=Asterisk PBX 1.8.2.2
  4023. c=IN IP4 208.211.92.75
  4024. t=0 0
  4025. m=audio 19710 RTP/AVP 0 101
  4026. a=rtpmap:0 PCMU/8000
  4027. a=rtpmap:101 telephone-event/8000
  4028. a=fmtp:101 0-16
  4029. a=silenceSupp:off - - - -
  4030. a=ptime:20
  4031. a=sendrecv
  4032.  
  4033. ---
  4034. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #119
  4035. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 212.7.117.61:48052
  4036. [Jan 26 08:42:35] DEBUG[23821] channel.c: Returning from native bridge, channels: SIP/dovid-0000000a, SIP/fpp-0000000b
  4037. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4038. Event: Unlink
  4039. Privilege: call,all
  4040. Channel1: SIP/dovid-0000000a
  4041. Channel2: SIP/fpp-0000000b
  4042. Uniqueid1: 1296049342.10
  4043. Uniqueid2: 1296049342.11
  4044. CallerID1: dovid
  4045. CallerID2: 10000009
  4046.  
  4047.  
  4048. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4049. Event: VarSet
  4050. Privilege: dialplan,all
  4051. Channel: SIP/dovid-0000000a
  4052. Variable: ANSWEREDTIME
  4053. Value: 10
  4054. Uniqueid: 1296049342.10
  4055.  
  4056.  
  4057. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4058. Event: VarSet
  4059. Privilege: dialplan,all
  4060. Channel: SIP/dovid-0000000a
  4061. Variable: DIALEDTIME
  4062. Value: 13
  4063. Uniqueid: 1296049342.10
  4064.  
  4065.  
  4066. [Jan 26 08:42:35] DEBUG[23821] cdr_mysql.c: Inserting a CDR record.
  4067. [Jan 26 08:42:35] DEBUG[23821] cdr_mysql.c: SQL command as follows: INSERT INTO asterisk_cdr (calldate,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags) VALUES ('2011-01-26 08:42:22','dovid','10000009','dovid','SIP/dovid-0000000a','SIP/fpp-0000000b','Dial','SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)','13','10','ANSWERED','3')
  4068. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '2011-01-26 08:42:22'
  4069. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '"dovid" <dovid>'
  4070. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'dovid'
  4071. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'SIP/dovid-0000000a'
  4072. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'SIP/fpp-0000000b'
  4073. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'Dial'
  4074. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)'
  4075. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '13'
  4076. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '10'
  4077. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'ANSWERED'
  4078. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is 'DOCUMENTATION'
  4079. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '(null)'
  4080. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '1296049342.10'
  4081. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '(null)'
  4082. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Function result is '(null)'
  4083. [Jan 26 08:42:35] DEBUG[23821] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-01-26 08:42:22','"dovid" <dovid>','dovid','SIP/dovid-0000000a','SIP/fpp-0000000b','Dial','SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)','13','10','ANSWERED','DOCUMENTATION','','1296049342.10','','')
  4084. [Jan 26 08:42:35] DEBUG[23821] channel.c: Hanging up channel 'SIP/fpp-0000000b'
  4085. [Jan 26 08:42:35] DEBUG[23821] chan_sip.c: Hangup call SIP/fpp-0000000b, SIP callid 730bfb20211d6c7a40e584041062e145@69.167.68.130
  4086. [Jan 26 08:42:35] DEBUG[23821] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd055c98'
  4087. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4088. Event: Hangup
  4089. Privilege: call,all
  4090. Channel: SIP/fpp-0000000b
  4091. Uniqueid: 1296049342.11
  4092. CallerIDNum: 10000009
  4093. CallerIDName: <unknown>
  4094. Cause: 16
  4095. Cause-txt: Normal Clearing
  4096.  
  4097.  
  4098. [Jan 26 08:42:35] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp
  4099. [Jan 26 08:42:35] DEBUG[23716] chan_sip.c: Checking device state for peer fpp
  4100. [Jan 26 08:42:35] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use)
  4101. [Jan 26 08:42:35] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1'
  4102. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4103. Event: VarSet
  4104. Privilege: dialplan,all
  4105. Channel: SIP/dovid-0000000a
  4106. Variable: DIALSTATUS
  4107. Value: ANSWER
  4108. Uniqueid: 1296049342.10
  4109.  
  4110.  
  4111. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4112. Event: Dial
  4113. Privilege: call,all
  4114. SubEvent: End
  4115. Channel: SIP/dovid-0000000a
  4116. UniqueID: 1296049342.10
  4117. DialStatus: ANSWER
  4118.  
  4119.  
  4120. [Jan 26 08:42:35] DEBUG[23821] app_dial.c: Exiting with DIALSTATUS=ANSWER.
  4121. [Jan 26 08:42:35] DEBUG[23821] pbx.c: Launching 'Playback'
  4122. [Jan 26 08:42:35] VERBOSE[23821] pbx.c: -- Executing [10000009@dovid:2] Playback("SIP/dovid-0000000a", "tt-monkeys") in new stack
  4123. [Jan 26 08:42:35] DEBUG[23750] manager.c: Examining event:
  4124. Event: Newexten
  4125. Privilege: dialplan,all
  4126. Channel: SIP/dovid-0000000a
  4127. Context: dovid
  4128. Extension: 10000009
  4129. Priority: 2
  4130. Application: Playback
  4131. AppData: tt-monkeys
  4132. Uniqueid: 1296049342.10
  4133.  
  4134.  
  4135. [Jan 26 08:42:35] DEBUG[23821] channel.c: Set channel SIP/dovid-0000000a to write format gsm
  4136. [Jan 26 08:42:35] DEBUG[23821] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
  4137. [Jan 26 08:42:35] DEBUG[23821] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
  4138. [Jan 26 08:42:35] DEBUG[23821] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xd050c00'
  4139. [Jan 26 08:42:35] DEBUG[23821] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  4140. [Jan 26 08:42:35] VERBOSE[23821] file.c: -- <SIP/dovid-0000000a> Playing 'tt-monkeys.gsm' (language 'en')
  4141. [Jan 26 08:42:36] DEBUG[23821] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  4142. [Jan 26 08:42:36] DEBUG[23821] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 212.7.117.61:53352
  4143. [Jan 26 08:42:36] VERBOSE[23737] chan_sip.c:
  4144. <--- SIP read from UDP:212.7.117.61:48052 --->
  4145. SIP/2.0 200 OK
  4146. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4103729c;rport=5060
  4147. Contact: <sip:dovid@212.7.117.61:48052>
  4148. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4149. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4150. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4151. CSeq: 103 INVITE
  4152. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  4153. Content-Type: application/sdp
  4154. User-Agent: eyeBeam release 1102q stamp 51814
  4155. Content-Length: 184
  4156.  
  4157. v=0
  4158. o=- 9 3 IN IP4 192.168.1.10
  4159. s=CounterPath eyeBeam 1.5
  4160. c=IN IP4 192.168.1.10
  4161. t=0 0
  4162. m=audio 53352 RTP/AVP 0 101
  4163. a=fmtp:101 0-15
  4164. a=rtpmap:101 telephone-event/8000
  4165. a=sendrecv
  4166. <------------->
  4167. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  4168. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4103729c;rport=5060
  4169. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 2 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  4170. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 3 [ 56]: To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4171. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 4 [ 66]: From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4172. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 5 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4173. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE
  4174. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  4175. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp
  4176. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 9 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  4177. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 10 [ 19]: Content-Length: 184
  4178. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Header 11 [ 0]:
  4179. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0
  4180. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 9 3 IN IP4 192.168.1.10
  4181. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5
  4182. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.10
  4183. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0
  4184. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 53352 RTP/AVP 0 101
  4185. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15
  4186. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  4187. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Body 8 [ 10]: a=sendrecv
  4188. [Jan 26 08:42:36] VERBOSE[23737] chan_sip.c: --- (11 headers 9 lines) ---
  4189. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking To) --From tag as083c547c --To-tag a23db027
  4190. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Acked pending invite 103
  4191. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #119
  4192. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Stopping retransmission on 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' of Request 103: Match Found
  4193. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4194. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  4195. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Call Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. responded to our reinvite without changing SDP version; ignoring SDP.
  4196. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Updating call counter for incoming call
  4197. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Strict routing enforced for session Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4198. [Jan 26 08:42:36] VERBOSE[23737] chan_sip.c: set_destination: Parsing <sip:dovid@212.7.117.61:48052> for address/port to send to
  4199. [Jan 26 08:42:36] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:48052' gives...
  4200. [Jan 26 08:42:36] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '48052'.
  4201. [Jan 26 08:42:36] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:48052
  4202. [Jan 26 08:42:36] VERBOSE[23737] chan_sip.c: Transmitting (NAT) to 212.7.117.61:48052:
  4203. ACK sip:dovid@212.7.117.61:48052 SIP/2.0
  4204. Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK185a0207;rport
  4205. Max-Forwards: 70
  4206. From: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4207. To: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4208. Contact: <sip:10000009@208.211.92.75:5060>
  4209. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4210. CSeq: 103 ACK
  4211. User-Agent: Asterisk PBX 1.8.2.2
  4212. Content-Length: 0
  4213.  
  4214.  
  4215. ---
  4216. [Jan 26 08:42:36] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:dov' onto UDP socket destined for 212.7.117.61:48052
  4217. [Jan 26 08:42:36] DEBUG[23750] manager.c: Examining event:
  4218. Event: VarSet
  4219. Privilege: dialplan,all
  4220. Channel: SIP/dovid-0000000a
  4221. Variable: ~HASH~SIP_CAUSE~SIP/dovid-0000000a~
  4222. Value: SIP 200 OK
  4223. Uniqueid: 1296049342.10
  4224.  
  4225.  
  4226. [Jan 26 08:42:36] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 176 bytes
  4227. [Jan 26 08:42:36] DEBUG[23750] manager.c: Examining event:
  4228. Event: RTCPReceived
  4229. Privilege: reporting,all
  4230. From 212.7.117.61:53353
  4231. PT: 200(Sender Report)
  4232. ReceptionReports: 1
  4233. SenderSSRC: 0
  4234. FractionLost: 0
  4235. PacketsLost: 0
  4236. HighestSequence: 40474
  4237. SequenceNumberCycles: 0
  4238. IAJitter: 15
  4239. LastSR: 41803.3758096384
  4240. DLSR: 1.1710(sec)
  4241. RTT: 79(sec)
  4242.  
  4243.  
  4244. [Jan 26 08:42:39] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 176 bytes
  4245. [Jan 26 08:42:39] DEBUG[23750] manager.c: Examining event:
  4246. Event: RTCPReceived
  4247. Privilege: reporting,all
  4248. From 212.7.117.61:53353
  4249. PT: 200(Sender Report)
  4250. ReceptionReports: 1
  4251. SenderSSRC: 0
  4252. FractionLost: 0
  4253. PacketsLost: 0
  4254. HighestSequence: 40474
  4255. SequenceNumberCycles: 0
  4256. IAJitter: 15
  4257. LastSR: 41803.3758096384
  4258. DLSR: 4.1760(sec)
  4259. RTT: 79(sec)
  4260.  
  4261.  
  4262. [Jan 26 08:42:40] DEBUG[23750] manager.c: Examining event:
  4263. Event: RTCPSent
  4264. Privilege: reporting,all
  4265. To 212.7.117.61:53353
  4266. OurSSRC: 1901277685
  4267. SentNTP: 1296049360.3483672576
  4268. SentRTP: 40000
  4269. SentPackets: 250
  4270. SentOctets: 40000
  4271. ReportBlock:
  4272. FractionLost: 6
  4273. CumulativeLoss: 6
  4274. IAJitter: 0.0015
  4275. TheirLastSR: 2740221902
  4276. DLSR: 1.1970 (sec)
  4277.  
  4278.  
  4279. [Jan 26 08:42:42] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 200 bytes
  4280. [Jan 26 08:42:42] DEBUG[23750] manager.c: Examining event:
  4281. Event: RTCPReceived
  4282. Privilege: reporting,all
  4283. From 212.7.117.61:53353
  4284. PT: 200(Sender Report)
  4285. ReceptionReports: 2
  4286. SenderSSRC: 0
  4287. FractionLost: 0
  4288. PacketsLost: 0
  4289. HighestSequence: 40474
  4290. SequenceNumberCycles: 0
  4291. IAJitter: 15
  4292. LastSR: 41803.3758096384
  4293. DLSR: 7.1810(sec)
  4294. RTT: 78(sec)
  4295.  
  4296.  
  4297. [Jan 26 08:42:45] VERBOSE[23737] chan_sip.c:
  4298. <--- SIP read from UDP:212.7.117.61:48052 --->
  4299.  
  4300.  
  4301. <------------->
  4302. [Jan 26 08:42:45] DEBUG[23737] chan_sip.c: Header 0 [ 0]:
  4303. [Jan 26 08:42:45] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 200 bytes
  4304. [Jan 26 08:42:45] DEBUG[23750] manager.c: Examining event:
  4305. Event: RTCPReceived
  4306. Privilege: reporting,all
  4307. From 212.7.117.61:53353
  4308. PT: 200(Sender Report)
  4309. ReceptionReports: 2
  4310. SenderSSRC: 0
  4311. FractionLost: 0
  4312. PacketsLost: 0
  4313. HighestSequence: 40474
  4314. SequenceNumberCycles: 0
  4315. IAJitter: 15
  4316. LastSR: 41803.3758096384
  4317. DLSR: 10.1850(sec)
  4318. RTT: 78(sec)
  4319.  
  4320.  
  4321. [Jan 26 08:42:45] DEBUG[23750] manager.c: Examining event:
  4322. Event: RTCPSent
  4323. Privilege: reporting,all
  4324. To 212.7.117.61:53353
  4325. OurSSRC: 1901277685
  4326. SentNTP: 1296049365.3487768576
  4327. SentRTP: 80000
  4328. SentPackets: 500
  4329. SentOctets: 80000
  4330. ReportBlock:
  4331. FractionLost: 8
  4332. CumulativeLoss: 14
  4333. IAJitter: 0.0015
  4334. TheirLastSR: 2740616167
  4335. DLSR: 0.1900 (sec)
  4336.  
  4337.  
  4338. [Jan 26 08:42:48] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 200 bytes
  4339. [Jan 26 08:42:48] DEBUG[23750] manager.c: Examining event:
  4340. Event: RTCPReceived
  4341. Privilege: reporting,all
  4342. From 212.7.117.61:53353
  4343. PT: 200(Sender Report)
  4344. ReceptionReports: 2
  4345. SenderSSRC: 0
  4346. FractionLost: 0
  4347. PacketsLost: 0
  4348. HighestSequence: 40474
  4349. SequenceNumberCycles: 0
  4350. IAJitter: 15
  4351. LastSR: 41803.3758096384
  4352. DLSR: 13.1910(sec)
  4353. RTT: 79(sec)
  4354.  
  4355.  
  4356. [Jan 26 08:42:50] DEBUG[23750] manager.c: Examining event:
  4357. Event: RTCPSent
  4358. Privilege: reporting,all
  4359. To 212.7.117.61:53353
  4360. OurSSRC: 1901277685
  4361. SentNTP: 1296049370.3484053504
  4362. SentRTP: 120000
  4363. SentPackets: 750
  4364. SentOctets: 120000
  4365. ReportBlock:
  4366. FractionLost: 5
  4367. CumulativeLoss: 19
  4368. IAJitter: 0.0030
  4369. TheirLastSR: 2740812775
  4370. DLSR: 2.1820 (sec)
  4371.  
  4372.  
  4373. [Jan 26 08:42:51] DEBUG[23821] res_rtp_asterisk.c: Got RTCP report of 160 bytes
  4374. [Jan 26 08:42:51] VERBOSE[23737] chan_sip.c:
  4375. <--- SIP read from UDP:212.7.117.61:48052 --->
  4376. BYE sip:10000009@208.211.92.75:5060 SIP/2.0
  4377. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-8153ef4dee314753-1---d8754z-;rport
  4378. Max-Forwards: 70
  4379. Contact: <sip:dovid@212.7.117.61:48052>
  4380. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4381. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4382. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4383. CSeq: 3 BYE
  4384. User-Agent: eyeBeam release 1102q stamp 51814
  4385. Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@208.211.92.75:5060",response="f33fff8adfbf992c5985cf1fab82c5e3",algorithm=MD5
  4386. Reason: SIP;description="User Hung Up"
  4387. Content-Length: 0
  4388.  
  4389. <------------->
  4390. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 0 [ 43]: BYE sip:10000009@208.211.92.75:5060 SIP/2.0
  4391. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-8153ef4dee314753-1---d8754z-;rport
  4392. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
  4393. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: <sip:dovid@212.7.117.61:48052>
  4394. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4395. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4396. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4397. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 3 BYE
  4398. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814
  4399. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 9 [168]: Authorization: Digest username="dovid",realm="asterisk",nonce="1da99604",uri="sip:10000009@208.211.92.75:5060",response="f33fff8adfbf992c5985cf1fab82c5e3",algorithm=MD5
  4400. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 10 [ 38]: Reason: SIP;description="User Hung Up"
  4401. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Header 11 [ 17]: Content-Length: 0
  4402. [Jan 26 08:42:51] VERBOSE[23737] chan_sip.c: --- (12 headers 0 lines) ---
  4403. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: = Looking for Call ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM. (Checking From) --From tag a23db027 --To-tag as083c547c
  4404. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
  4405. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Initializing initreq for method BYE - callid Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4406. [Jan 26 08:42:51] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:48052' gives...
  4407. [Jan 26 08:42:51] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '48052'.
  4408. [Jan 26 08:42:51] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:48052 (NAT)
  4409. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Setting SIP_ALREADYGONE on dialog Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4410. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4411. Event: VarSet
  4412. Privilege: dialplan,all
  4413. Channel: SIP/dovid-0000000a
  4414. Variable: RTPAUDIOQOS
  4415. Value: ssrc=1901277685;themssrc=1259751702;lp=19;rxjitter=0.001763;rxcount=742;txjitter=0.000000;txcount=779;rlp=0;rtt=0.079000
  4416. Uniqueid: 1296049342.10
  4417.  
  4418.  
  4419. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4420. Event: VarSet
  4421. Privilege: dialplan,all
  4422. Channel: SIP/dovid-0000000a
  4423. Variable: RTPAUDIOQOSJITTER
  4424. Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
  4425. Uniqueid: 1296049342.10
  4426.  
  4427.  
  4428. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4429. Event: VarSet
  4430. Privilege: dialplan,all
  4431. Channel: SIP/dovid-0000000a
  4432. Variable: RTPAUDIOQOSLOSS
  4433. Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
  4434. Uniqueid: 1296049342.10
  4435.  
  4436.  
  4437. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4438. Event: VarSet
  4439. Privilege: dialplan,all
  4440. Channel: SIP/dovid-0000000a
  4441. Variable: RTPAUDIOQOSRTT
  4442. Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
  4443. Uniqueid: 1296049342.10
  4444.  
  4445.  
  4446. [Jan 26 08:42:51] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  4447. [Jan 26 08:42:51] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog 'Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.' in 32000 ms (Method: BYE)
  4448. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Received bye, issuing owner hangup
  4449. [Jan 26 08:42:51] VERBOSE[23737] chan_sip.c:
  4450. <--- Transmitting (NAT) to 212.7.117.61:48052 --->
  4451. SIP/2.0 200 OK
  4452. Via: SIP/2.0/UDP 192.168.1.10:48052;branch=z9hG4bK-d8754z-8153ef4dee314753-1---d8754z-;received=212.7.117.61;rport=48052
  4453. From: "dovid"<sip:dovid@mypbx.mydomain.com>;tag=a23db027
  4454. To: "10000009"<sip:10000009@mypbx.mydomain.com>;tag=as083c547c
  4455. Call-ID: Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4456. CSeq: 3 BYE
  4457. Server: Asterisk PBX 1.8.2.2
  4458. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  4459. Supported: replaces, timer
  4460. Content-Length: 0
  4461.  
  4462.  
  4463. <------------>
  4464. [Jan 26 08:42:51] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.7.117.61:48052
  4465. [Jan 26 08:42:51] WARNING[23821] file.c: Failed to write frame
  4466. [Jan 26 08:42:51] DEBUG[23821] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  4467. [Jan 26 08:42:51] DEBUG[23821] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  4468. [Jan 26 08:42:51] DEBUG[23821] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  4469. [Jan 26 08:42:51] DEBUG[23821] channel.c: Set channel SIP/dovid-0000000a to write format ulaw
  4470. [Jan 26 08:42:51] DEBUG[23821] pbx.c: Spawn extension (dovid,10000009,2) exited non-zero on 'SIP/dovid-0000000a'
  4471. [Jan 26 08:42:51] VERBOSE[23821] pbx.c: == Spawn extension (dovid, 10000009, 2) exited non-zero on 'SIP/dovid-0000000a'
  4472. [Jan 26 08:42:51] DEBUG[23821] channel.c: Soft-Hanging up channel 'SIP/dovid-0000000a'
  4473. [Jan 26 08:42:51] DEBUG[23821] channel.c: Hanging up channel 'SIP/dovid-0000000a'
  4474. [Jan 26 08:42:51] DEBUG[23821] chan_sip.c: Hangup call SIP/dovid-0000000a, SIP callid Y2YzMjc2MzllZGNiMmY1NDBmNDhiYWJkOGE5ZTMwOWM.
  4475. [Jan 26 08:42:51] DEBUG[23821] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd050c00'
  4476. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4477. Event: VarSet
  4478. Privilege: dialplan,all
  4479. Channel: SIP/dovid-0000000a
  4480. Variable: PLAYBACKSTATUS
  4481. Value: SUCCESS
  4482. Uniqueid: 1296049342.10
  4483.  
  4484.  
  4485. [Jan 26 08:42:51] DEBUG[23750] manager.c: Examining event:
  4486. Event: Hangup
  4487. Privilege: call,all
  4488. Channel: SIP/dovid-0000000a
  4489. Uniqueid: 1296049342.10
  4490. CallerIDNum: dovid
  4491. CallerIDName: dovid
  4492. Cause: 16
  4493. Cause-txt: Normal Clearing
  4494.  
  4495.  
  4496. [Jan 26 08:42:51] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid
  4497. [Jan 26 08:42:51] DEBUG[23716] chan_sip.c: Checking device state for peer dovid
  4498. [Jan 26 08:42:51] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use)
  4499. [Jan 26 08:42:51] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1'
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