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  1. [Jul 18 16:12:06] VERBOSE[24157] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 18 16:12:06] DEBUG[24157] config.c: Parsing /etc/asterisk/logger.conf
  2. [Jul 18 16:12:06] VERBOSE[24157] config.c: == Found
  3. [Jul 18 16:12:06] VERBOSE[24157] logger.c: Asterisk Event Logger restarted
  4. [Jul 18 16:12:06] VERBOSE[24157] logger.c: Asterisk Queue Logger restarted
  5. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
  6. <--- SIP read from UDP:192.168.20.1:5060 --->
  7. INVITE sip:[email protected] SIP/2.0
  8. Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9
  9. From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  10. To: <sip:[email protected]>
  11. Call-ID: [email protected]
  12. CSeq: 1798690992 INVITE
  13. Contact: "Mushtaq" <sip:[email protected]:5060;transport=udp>
  14. Max-Forwards: 70
  15. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
  16. Content-Type: application/sdp
  17. Supported: timer
  18. Content-Length: 275
  19.  
  20. v=0
  21. o=UserA 1459024281 4215621787 IN IP4 192.168.20.1
  22. s=Session SDP
  23. c=IN IP4 192.168.20.1
  24. t=0 0
  25. m=audio 49154 RTP/AVP 8 0 18 101
  26. a=rtpmap:8 PCMA/8000
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:18 G729/8000
  29. a=fmtp:18 annexb=no
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-15
  32.  
  33. <------------->
  34. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 0 [ 35]: INVITE sip:[email protected] SIP/2.0
  35. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9
  36. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 2 [ 59]: From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  37. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 3 [ 26]: To: <sip:[email protected]>
  38. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: [email protected]
  39. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 5 [ 23]: CSeq: 1798690992 INVITE
  40. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 6 [ 60]: Contact: "Mushtaq" <sip:[email protected]:5060;transport=udp>
  41. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
  42. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 8 [ 69]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
  43. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp
  44. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 10 [ 16]: Supported: timer
  45. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 11 [ 19]: Content-Length: 275
  46. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 12 [ 0]:
  47. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 0 [ 3]: v=0
  48. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 1 [ 49]: o=UserA 1459024281 4215621787 IN IP4 192.168.20.1
  49. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 2 [ 13]: s=Session SDP
  50. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.20.1
  51. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 4 [ 5]: t=0 0
  52. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 5 [ 32]: m=audio 49154 RTP/AVP 8 0 18 101
  53. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000
  54. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
  55. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000
  56. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no
  57. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000
  58. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15
  59. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: --- (12 headers 12 lines) ---
  60. [Jul 18 16:13:15] DEBUG[27072] acl.c: Found IP address for this socket
  61. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.40:5060
  62. [Jul 18 16:13:15] VERBOSE[27072] netsock.c: == Using SIP RTP CoS mark 5
  63. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting NAT on RTP to On
  64. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Allocating new SIP dialog for [email protected] - INVITE (With RTP)
  65. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  66. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Begin: parsing SIP "Supported: timer"
  67. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Found SIP option: -timer-
  68. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Matched SIP option: timer
  69. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Sending to 192.168.20.1 : 5060 (NAT)
  70. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Initializing initreq for method INVITE - callid [email protected]
  71. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Using INVITE request as basis request - [email protected]
  72. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found peer 'avaya' for '254' from 192.168.20.1:5060
  73. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting NAT on RTP to On
  74. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  75. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP o=UserA 1459024281 4215621787 IN IP4 192.168.20.1... UNSUPPORTED.
  76. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP s=Session SDP... UNSUPPORTED.
  77. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.20.1... OK.
  78. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  79. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 8
  80. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 0
  81. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 18
  82. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 101
  83. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format PCMA for ID 8
  84. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
  85. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format PCMU for ID 0
  86. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  87. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format G729 for ID 18
  88. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
  89. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
  90. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format telephone-event for ID 101
  91. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  92. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
  93. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
  94. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  95. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Peer audio RTP is at port 192.168.20.1:49154
  96. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: We're settling with these formats: 0x10c (ulaw|alaw|g729)
  97. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Checking SIP call limits for device
  98. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Updating call counter for incoming call
  99. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Looking for 300 in access (domain 192.168.1.40)
  100. [Jul 18 16:13:15] DEBUG[27072] frame.c: Could not find preferred codec - Going for the best codec
  101. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Our native formats are 0x4 (ulaw)
  102. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Joint capabilities are 0x10c (ulaw|alaw|g729)
  103. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Our capabilities are 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14)
  104. [Jul 18 16:13:15] DEBUG[27072] frame.c: Could not find preferred codec - Going for the best codec
  105. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
  106. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: This channel will not be able to handle video.
  107. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: build_route: Contact hop: "Mushtaq" <sip:[email protected]:5060;transport=udp>
  108. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: list_route: hop: <sip:[email protected]:5060;transport=udp>
  109. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: SIP/avaya-000000d5: New call is still down.... Trying...
  110. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
  111. <--- Transmitting (NAT) to 192.168.20.1:5060 --->
  112. SIP/2.0 100 Trying
  113. Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9;received=192.168.20.1;rport=5060
  114. From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  115. To: <sip:[email protected]>
  116. Call-ID: [email protected]
  117. CSeq: 1798690992 INVITE
  118. Server: Asterisk PBX 1.6.2.22
  119. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  120. Supported: replaces, timer
  121. Contact: <sip:[email protected]>
  122. Content-Length: 0
  123.  
  124.  
  125. <------------>
  126. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.20.1:5060
  127. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
  128. [Jul 18 16:13:15] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
  129. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
  130. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
  131. [Jul 18 16:13:15] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  132. [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'Answer'
  133. [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:1] Answer("SIP/avaya-000000d5", "") in new stack
  134. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
  135. [Jul 18 16:13:15] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
  136. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
  137. [Jul 18 16:13:15] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
  138. [Jul 18 16:13:15] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  139. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: SIP answering channel: SIP/avaya-000000d5
  140. [Jul 18 16:13:15] DEBUG[25493] rtp.c: Setting the marker bit due to a source update
  141. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Setting framing from config on incoming call
  142. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: True Text flag: True
  143. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  144. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Audio is at 192.168.1.40 port 11258
  145. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  146. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  147. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x100 (g729) to SDP
  148. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  149. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: -- Done with adding codecs to SDP
  150. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Done building SDP. Settling with this capability: 0x10c (ulaw|alaw|g729)
  151. [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c:
  152. <--- Reliably Transmitting (NAT) to 192.168.20.1:5060 --->
  153. SIP/2.0 200 OK
  154. Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9;received=192.168.20.1;rport=5060
  155. From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  156. To: <sip:[email protected]>;tag=as09decdab
  157. Call-ID: [email protected]
  158. CSeq: 1798690992 INVITE
  159. Server: Asterisk PBX 1.6.2.22
  160. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  161. Supported: replaces, timer
  162. Contact: <sip:[email protected]>
  163. Content-Type: application/sdp
  164. Content-Length: 306
  165.  
  166. v=0
  167. o=root 805920992 805920992 IN IP4 192.168.1.40
  168. s=Asterisk PBX 1.6.2.22
  169. c=IN IP4 192.168.1.40
  170. t=0 0
  171. m=audio 11258 RTP/AVP 0 8 18 101
  172. a=rtpmap:0 PCMU/8000
  173. a=rtpmap:8 PCMA/8000
  174. a=rtpmap:18 G729/8000
  175. a=fmtp:18 annexb=no
  176. a=rtpmap:101 telephone-event/8000
  177. a=fmtp:101 0-16
  178. a=ptime:20
  179. a=sendrecv
  180.  
  181. <------------>
  182. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25179
  183. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.20.1:5060
  184. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
  185. <--- SIP read from UDP:192.168.20.1:5060 --->
  186. ACK sip:[email protected] SIP/2.0
  187. Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK2295d3be4d069b089fe3bba9eb3606b3
  188. From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  189. To: <sip:[email protected]>;tag=as09decdab
  190. Call-ID: [email protected]
  191. CSeq: 1798690992 ACK
  192. Max-Forwards: 70
  193. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
  194. Content-Length: 0
  195.  
  196.  
  197. <------------->
  198. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 0 [ 32]: ACK sip:[email protected] SIP/2.0
  199. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK2295d3be4d069b089fe3bba9eb3606b3
  200. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 2 [ 59]: From: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  201. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 3 [ 41]: To: <sip:[email protected]>;tag=as09decdab
  202. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: [email protected]
  203. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 5 [ 20]: CSeq: 1798690992 ACK
  204. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70
  205. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 7 [ 69]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
  206. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 8 [ 17]: Content-Length: 0
  207. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 9 [ 0]:
  208. [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: --- (9 headers 0 lines) ---
  209. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  210. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25179
  211. [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Stopping retransmission on '[email protected]' of Response 1798690992: Match Found
  212. [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Oooh, format changed to 8 alaw
  213. [Jul 18 16:13:15] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to read format ulaw
  214. [Jul 18 16:13:15] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
  215. [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'Set'
  216. [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:2] Set("SIP/avaya-000000d5", "[email protected]") in new stack
  217. [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'AGI'
  218. [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:3] AGI("SIP/avaya-000000d5", "agi://192.168.1.41/testtreatment,inbound") in new stack
  219. [Jul 18 16:13:15] DEBUG[25493] res_agi.c: Wow, connected!
  220. [Jul 18 16:13:15] NOTICE[25493] channel.c: Dropping incompatible voice frame on SIP/avaya-000000d5 of format ulaw since our native format has changed to 0x8 (alaw)
  221. [Jul 18 16:13:16] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
  222. [Jul 18 16:13:16] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
  223. [Jul 18 16:13:16] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/Welcome' (escape_digits=) (sample_offset 0)
  224. [Jul 18 16:13:16] DEBUG[25493] rtp.c: Ooh, format changed from unknown to alaw
  225. [Jul 18 16:13:16] DEBUG[25493] rtp.c: Created smoother: format: 8 ms: 20 len: 160
  226. [Jul 18 16:13:16] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  227. [Jul 18 16:13:19] DEBUG[25493] rtp.c: Got RTCP report of 88 bytes
  228. [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (73 requested / 73 actual) timer ticks per second
  229. [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  230. [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  231. [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  232. [Jul 18 16:13:20] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
  233. [Jul 18 16:13:20] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
  234. [Jul 18 16:13:20] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
  235. [Jul 18 16:13:20] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/LanguageSelection' (escape_digits=0123456789*#) (sample_offset 0)
  236. [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  237. [Jul 18 16:13:23] DEBUG[25493] rtp.c: Got RTCP report of 88 bytes
  238. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  239. [Jul 18 16:13:26] DEBUG[25493] rtp.c: Sending dtmf: 49 (1), at 192.168.20.1
  240. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  241. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  242. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  243. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  244. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  245. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  246. [Jul 18 16:13:26] DEBUG[25493] rtp.c: Sending dtmf: 49 (1), at 192.168.20.1
  247. [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  248. [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  249. [Jul 18 16:13:26] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
  250. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  251. [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
  252. [Jul 18 16:13:26] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
  253. [Jul 18 16:13:26] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
  254. [Jul 18 16:13:26] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/E_MainMenu' (escape_digits=0123456789*#) (sample_offset 0)
  255. [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  256. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  257. [Jul 18 16:13:29] DEBUG[25493] rtp.c: Sending dtmf: 48 (0), at 192.168.20.1
  258. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  259. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  260. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  261. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  262. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  263. [Jul 18 16:13:29] DEBUG[25493] rtp.c: Sending dtmf: 48 (0), at 192.168.20.1
  264. [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  265. [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  266. [Jul 18 16:13:29] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
  267. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  268. [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
  269. [Jul 18 16:13:29] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
  270. [Jul 18 16:13:29] DEBUG[25493] format_wav.c: Skipping unknown block 'LIST'
  271. [Jul 18 16:13:29] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/E_TransferToAgent' (escape_digits=) (sample_offset 0)
  272. [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
  273. [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (52 requested / 52 actual) timer ticks per second
  274. [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  275. [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  276. [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
  277. [Jul 18 16:13:34] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
  278. [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- AGI Script Executing Application: (Transfer) Options: (SIP/avaya/6000)
  279. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: SIP transfer of [email protected] to avaya/6000
  280. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Strict routing enforced for session [email protected]
  281. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
  282. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: set destination to 192.168.20.1, port 5060
  283. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Reliably Transmitting (NAT) to 192.168.20.1:5060:
  284. REFER sip:[email protected]:5060;transport=udp SIP/2.0
  285. Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
  286. Max-Forwards: 70
  287. From: <sip:[email protected]>;tag=as09decdab
  288. To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  289. Contact: <sip:[email protected]>
  290. Call-ID: [email protected]
  291. CSeq: 102 REFER
  292. User-Agent: Asterisk PBX 1.6.2.22
  293. Refer-To: <sip:avaya/[email protected]>
  294. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  295. Supported: replaces, timer
  296. Referred-By: <sip:[email protected]>
  297. Content-Length: 0
  298.  
  299.  
  300. ---
  301. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25181
  302. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Trying to put 'REFER sip:2' onto UDP socket destined for 192.168.20.1:5060
  303. [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- AGI Script Executing Application: (NoOp) Options: (${TRANSFERSTATUS})
  304. [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- <SIP/avaya-000000d5>AGI Script agi://192.168.1.41/testtreatment completed, returning 0
  305. [Jul 18 16:13:34] VERBOSE[25493] pbx.c: -- Auto fallthrough, channel 'SIP/avaya-000000d5' status is 'UNKNOWN'
  306. [Jul 18 16:13:34] DEBUG[25493] channel.c: Soft-Hanging up channel 'SIP/avaya-000000d5'
  307. [Jul 18 16:13:34] DEBUG[25493] channel.c: Hanging up channel 'SIP/avaya-000000d5'
  308. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Hangup call SIP/avaya-000000d5, SIP callid [email protected]
  309. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: ACK)
  310. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Strict routing enforced for session [email protected]
  311. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
  312. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: set destination to 192.168.20.1, port 5060
  313. [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Reliably Transmitting (NAT) to 192.168.20.1:5060:
  314. BYE sip:[email protected]:5060;transport=udp SIP/2.0
  315. Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
  316. Max-Forwards: 70
  317. From: <sip:[email protected]>;tag=as09decdab
  318. To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  319. Call-ID: [email protected]
  320. CSeq: 103 BYE
  321. User-Agent: Asterisk PBX 1.6.2.22
  322. X-Asterisk-HangupCause: Unknown
  323. X-Asterisk-HangupCauseCode: 0
  324. Content-Length: 0
  325.  
  326.  
  327. ---
  328. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25183
  329. [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Trying to put 'BYE sip:254' onto UDP socket destined for 192.168.20.1:5060
  330. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c:
  331. <--- SIP read from UDP:192.168.20.1:5060 --->
  332. SIP/2.0 481 Dialog/Transaction Does Not Exist
  333. Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
  334. From: <sip:[email protected]>;tag=as09decdab
  335. To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  336. Call-ID: [email protected]
  337. CSeq: 102 REFER
  338. Supported: timer
  339. Content-Length: 0
  340.  
  341.  
  342. <------------->
  343. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 0 [ 45]: SIP/2.0 481 Dialog/Transaction Does Not Exist
  344. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
  345. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 2 [ 43]: From: <sip:[email protected]>;tag=as09decdab
  346. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 3 [ 57]: To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  347. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: [email protected]
  348. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 5 [ 15]: CSeq: 102 REFER
  349. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Supported: timer
  350. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  351. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 8 [ 0]:
  352. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: --- (8 headers 0 lines) ---
  353. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25181
  354. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
  355. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: SIP Response message for INCOMING dialog REFER arrived
  356. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Remote host can't match request REFER to call '[email protected]'. Giving up
  357. [Jul 18 16:13:34] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
  358. [Jul 18 16:13:34] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
  359. [Jul 18 16:13:34] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
  360. [Jul 18 16:13:34] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
  361. [Jul 18 16:13:34] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  362. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c:
  363. <--- SIP read from UDP:192.168.20.1:5060 --->
  364. SIP/2.0 200 Ok
  365. Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
  366. From: <sip:[email protected]>;tag=as09decdab
  367. To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  368. Call-ID: [email protected]
  369. CSeq: 103 BYE
  370. Supported: timer
  371. Content-Length: 0
  372.  
  373.  
  374. <------------->
  375. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok
  376. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
  377. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 2 [ 43]: From: <sip:[email protected]>;tag=as09decdab
  378. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 3 [ 57]: To: "Mushtaq" <sip:[email protected]>;tag=2d5dbb8afb9c01c0
  379. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: [email protected]
  380. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE
  381. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Supported: timer
  382. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  383. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 8 [ 0]:
  384. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: --- (8 headers 0 lines) ---
  385. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25183
  386. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
  387. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
  388. [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Destroying SIP dialog [email protected]
  389. [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: ACK
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