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console-trunk-r319661.log

level7
May 20th, 2011
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  1. [May 20 13:49:49] VERBOSE[5717] chan_sip.c:
  2. <--- SIP read from UDP:192.168.1.52:5060 --->
  3. INVITE sip:501@eu.sip.l7dev.co.cc SIP/2.0
  4. Record-Route: <sip:192.168.1.52;lr=on;ftag=jdnch>
  5. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.0
  6. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKhfuakcot
  7. Max-Forwards: 69
  8. To: <sip:501@eu.sip.l7dev.co.cc>
  9. From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  10. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  11. CSeq: 638 INVITE
  12. Contact: <sip:10022@192.168.1.23:5060>
  13. Content-Type: application/sdp
  14. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  15. Supported: replaces,norefersub,100rel
  16. User-Agent: Twinkle/1.4.2
  17. Content-Length: 308
  18.  
  19. v=0
  20. o=twinkle 1346033228 944464517 IN IP4 192.168.1.23
  21. s=-
  22. c=IN IP4 192.168.1.23
  23. t=0 0
  24. m=audio 8000 RTP/AVP 98 97 8 0 3 101
  25. a=rtpmap:98 speex/16000
  26. a=rtpmap:97 speex/8000
  27. a=rtpmap:8 PCMA/8000
  28. a=rtpmap:0 PCMU/8000
  29. a=rtpmap:3 GSM/8000
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-15
  32. a=ptime:20
  33. <------------->
  34. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 0 [ 41]: INVITE sip:501@eu.sip.l7dev.co.cc SIP/2.0
  35. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 1 [ 49]: Record-Route: <sip:192.168.1.52;lr=on;ftag=jdnch>
  36. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 2 [ 59]: Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.0
  37. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 3 [ 85]: Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKhfuakcot
  38. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 4 [ 16]: Max-Forwards: 69
  39. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 5 [ 32]: To: <sip:501@eu.sip.l7dev.co.cc>
  40. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 6 [ 54]: From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  41. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 7 [ 39]: Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  42. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 8 [ 16]: CSeq: 638 INVITE
  43. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 9 [ 38]: Contact: <sip:10022@192.168.1.23:5060>
  44. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp
  45. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 11 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  46. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 12 [ 37]: Supported: replaces,norefersub,100rel
  47. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 13 [ 25]: User-Agent: Twinkle/1.4.2
  48. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 14 [ 19]: Content-Length: 308
  49. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 17 [ 0]:
  50. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 0 [ 3]: v=0
  51. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 1 [ 50]: o=twinkle 1346033228 944464517 IN IP4 192.168.1.23
  52. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 2 [ 3]: s=-
  53. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.23
  54. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 4 [ 5]: t=0 0
  55. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 5 [ 36]: m=audio 8000 RTP/AVP 98 97 8 0 3 101
  56. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 6 [ 23]: a=rtpmap:98 speex/16000
  57. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 7 [ 22]: a=rtpmap:97 speex/8000
  58. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000
  59. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000
  60. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000
  61. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000
  62. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15
  63. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Body 13 [ 10]: a=ptime:20
  64. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: --- (17 headers 14 lines) ---
  65. [May 20 13:49:49] DEBUG[5717] chan_sip.c: = Looking for Call ID: ahmejjysyyetcgl@chris-ubuntu10 (Checking From) --From tag jdnch --To-tag
  66. [May 20 13:49:49] DEBUG[5717] acl.c: For destination '192.168.1.52', our source address is '192.168.1.74'.
  67. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.74:5060
  68. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Allocating new SIP dialog for ahmejjysyyetcgl@chris-ubuntu10 - INVITE (No RTP)
  69. [May 20 13:49:49] DEBUG[5717] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  70. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel"
  71. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Found SIP option: -replaces-
  72. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Matched SIP option: replaces
  73. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Found SIP option: -norefersub-
  74. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Matched SIP option: norefersub
  75. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Found SIP option: -100rel-
  76. [May 20 13:49:49] DEBUG[5717] sip/reqresp_parser.c: Matched SIP option: 100rel
  77. [May 20 13:49:49] DEBUG[5717] netsock2.c: Splitting '192.168.1.52' gives...
  78. [May 20 13:49:49] DEBUG[5717] netsock2.c: ...host '192.168.1.52' and port '(null)'.
  79. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Sending to 192.168.1.52:5060 (NAT)
  80. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Initializing initreq for method INVITE - callid ahmejjysyyetcgl@chris-ubuntu10
  81. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Using INVITE request as basis request - ahmejjysyyetcgl@chris-ubuntu10
  82. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found peer 'OpenSER' for '10022' from 192.168.1.52:5060
  83. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xdfdb98'
  84. [May 20 13:49:49] DEBUG[5717] res_rtp_asterisk.c: Allocated port 13328 for RTP instance '0xdfdb98'
  85. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: RTP instance '0xdfdb98' is setup and ready to go
  86. [May 20 13:49:49] DEBUG[5717] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xdfdb98'
  87. [May 20 13:49:49] VERBOSE[5717] netsock2.c: == Using SIP RTP CoS mark 5
  88. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Setting NAT on RTP to On
  89. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  90. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing session-level SDP o=twinkle 1346033228 944464517 IN IP4 192.168.1.23... UNSUPPORTED.
  91. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED.
  92. [May 20 13:49:49] DEBUG[5717] netsock2.c: Splitting '192.168.1.23' gives...
  93. [May 20 13:49:49] DEBUG[5717] netsock2.c: ...host '192.168.1.23' and port '(null)'.
  94. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.23... OK.
  95. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  96. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 98
  97. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 98 based on m type on 0x406451d0
  98. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 97
  99. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 97 based on m type on 0x406451d0
  100. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 8
  101. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 8 based on m type on 0x406451d0
  102. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 0
  103. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 0 based on m type on 0x406451d0
  104. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 3
  105. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 3 based on m type on 0x406451d0
  106. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found RTP audio format 101
  107. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Setting payload 101 based on m type on 0x406451d0
  108. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format speex for ID 98
  109. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 speex/16000... OK.
  110. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format speex for ID 97
  111. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 speex/8000... OK.
  112. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format PCMA for ID 8
  113. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
  114. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format PCMU for ID 0
  115. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  116. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format GSM for ID 3
  117. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
  118. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Found audio description format telephone-event for ID 101
  119. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  120. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
  121. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  122. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 0 on 0x406451d0
  123. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 3 on 0x406451d0
  124. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 8 on 0x406451d0
  125. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 97 on 0x406451d0
  126. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 98 on 0x406451d0
  127. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Incorporating payload 101 on 0x406451d0
  128. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|speex|g722), peer - audio=(gsm|ulaw|alaw|speex|speex16)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|speex)
  129. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  130. [May 20 13:49:49] DEBUG[5717] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xdfdb98'
  131. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Peer audio RTP is at port 192.168.1.23:8000
  132. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 0 from 0x406451d0 to 0xdfdd60
  133. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 3 from 0x406451d0 to 0xdfdd60
  134. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 8 from 0x406451d0 to 0xdfdd60
  135. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 97 from 0x406451d0 to 0xdfdd60
  136. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 98 from 0x406451d0 to 0xdfdd60
  137. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Copying payload 101 from 0x406451d0 to 0xdfdd60
  138. [May 20 13:49:49] DEBUG[5717] chan_sip.c: We're settling with these formats: (gsm|ulaw|alaw|speex)
  139. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Checking SIP call limits for device
  140. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Updating call counter for incoming call
  141. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Looking for 501 in common (domain eu.sip.l7dev.co.cc)
  142. [May 20 13:49:49] DEBUG[5717] chan_sip.c: *** Our native formats are (alaw)
  143. [May 20 13:49:49] DEBUG[5717] chan_sip.c: *** Joint capabilities are (gsm|ulaw|alaw|speex)
  144. [May 20 13:49:49] DEBUG[5717] chan_sip.c: *** Our capabilities are (g723|gsm|ulaw|alaw|g726|speex|g722)
  145. [May 20 13:49:49] DEBUG[5717] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw
  146. [May 20 13:49:49] DEBUG[5717] chan_sip.c: This channel will not be able to handle video.
  147. [May 20 13:49:49] DEBUG[5717] chan_sip.c: build_route: Record-Route hop: <sip:192.168.1.52;lr=on;ftag=jdnch>
  148. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: list_route: hop: <sip:192.168.1.52;lr=on;ftag=jdnch>
  149. [May 20 13:49:49] DEBUG[5717] chan_sip.c: SIP/OpenSER-00000000: New call is still down.... Trying...
  150. [May 20 13:49:49] VERBOSE[5717] chan_sip.c:
  151. <--- Transmitting (NAT) to 192.168.1.52:5060 --->
  152. SIP/2.0 100 Trying
  153.  
  154. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.0;received=192.168.1.52;rport=5060
  155.  
  156. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKhfuakcot
  157.  
  158. Record-Route: <sip:192.168.1.52;lr=on;ftag=jdnch>
  159.  
  160. From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  161.  
  162. To: <sip:501@eu.sip.l7dev.co.cc>
  163.  
  164. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  165.  
  166. CSeq: 638 INVITE
  167.  
  168. Server: Media GW 1
  169.  
  170. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  171.  
  172. Supported: replaces, timer
  173.  
  174. Contact: <sip:501@192.168.1.74:5060>
  175.  
  176. Content-Length: 0
  177.  
  178.  
  179.  
  180.  
  181. <------------>
  182. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.52:5060
  183. [May 20 13:49:49] DEBUG[5710] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  184. [May 20 13:49:49] DEBUG[5710] chan_sip.c: Checking device state for peer OpenSER
  185. [May 20 13:49:49] DEBUG[5710] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  186. [May 20 13:49:49] DEBUG[5710] devicestate.c: device 'SIP/OpenSER' state '1'
  187. [May 20 13:49:49] DEBUG[5719] pbx.c: Launching 'NoOp'
  188. [May 20 13:49:49] VERBOSE[5719] pbx.c: -- Executing [501@common:1] NoOp("SIP/OpenSER-00000000", "#### [common] ####") in new stack
  189. [May 20 13:49:49] DEBUG[5719] pbx.c: Launching 'Answer'
  190. [May 20 13:49:49] VERBOSE[5719] pbx.c: -- Executing [501@common:2] Answer("SIP/OpenSER-00000000", "") in new stack
  191. [May 20 13:49:49] DEBUG[5719] chan_sip.c: SIP answering channel: SIP/OpenSER-00000000
  192. [May 20 13:49:49] DEBUG[5719] res_rtp_asterisk.c: Setting the marker bit due to a source update
  193. [May 20 13:49:49] DEBUG[5719] chan_sip.c: Setting framing from config on incoming call
  194. [May 20 13:49:49] DEBUG[5719] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|speex) Video flag: True Text flag: True
  195. [May 20 13:49:49] DEBUG[5719] chan_sip.c: ** Our prefcodec: (nothing)
  196. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Audio is at 5060
  197. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Adding codec 100004 (alaw) to SDP
  198. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Adding codec 100003 (ulaw) to SDP
  199. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Adding codec 100002 (gsm) to SDP
  200. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Adding codec 100009 (speex) to SDP
  201. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  202. [May 20 13:49:49] DEBUG[5719] chan_sip.c: -- Done with adding codecs to SDP
  203. [May 20 13:49:49] DEBUG[5719] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|speex)
  204. [May 20 13:49:49] VERBOSE[5719] chan_sip.c:
  205. <--- Reliably Transmitting (NAT) to 192.168.1.52:5060 --->
  206. SIP/2.0 200 OK
  207.  
  208. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.0;received=192.168.1.52;rport=5060
  209.  
  210. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKhfuakcot
  211.  
  212. Record-Route: <sip:192.168.1.52;lr=on;ftag=jdnch>
  213.  
  214. From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  215.  
  216. To: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  217.  
  218. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  219.  
  220. CSeq: 638 INVITE
  221.  
  222. Server: Media GW 1
  223.  
  224. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  225.  
  226. Supported: replaces, timer
  227.  
  228. Contact: <sip:501@192.168.1.74:5060>
  229.  
  230. Content-Type: application/sdp
  231.  
  232. Content-Length: 325
  233.  
  234.  
  235.  
  236. v=0
  237.  
  238. o=root 191413774 191413774 IN IP4 192.168.1.74
  239.  
  240. s=Media GW 1
  241.  
  242. c=IN IP4 192.168.1.74
  243.  
  244. t=0 0
  245.  
  246. m=audio 13328 RTP/AVP 8 0 3 97 101
  247.  
  248. a=rtpmap:8 PCMA/8000
  249.  
  250. a=rtpmap:0 PCMU/8000
  251.  
  252. a=rtpmap:3 GSM/8000
  253.  
  254. a=rtpmap:97 speex/8000
  255.  
  256. a=rtpmap:101 telephone-event/8000
  257.  
  258. a=fmtp:101 0-16
  259.  
  260. a=silenceSupp:off - - - -
  261.  
  262. a=ptime:20
  263.  
  264. a=sendrecv
  265.  
  266.  
  267. <------------>
  268. [May 20 13:49:49] DEBUG[5719] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6
  269. [May 20 13:49:49] DEBUG[5719] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.52:5060
  270. [May 20 13:49:49] DEBUG[5710] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  271. [May 20 13:49:49] DEBUG[5710] chan_sip.c: Checking device state for peer OpenSER
  272. [May 20 13:49:49] DEBUG[5710] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  273. [May 20 13:49:49] DEBUG[5710] devicestate.c: device 'SIP/OpenSER' state '1'
  274. [May 20 13:49:49] VERBOSE[5717] chan_sip.c:
  275. <--- SIP read from UDP:192.168.1.52:5060 --->
  276. ACK sip:501@192.168.1.74:5060 SIP/2.0
  277. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.2
  278. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKfbvogxma
  279. Max-Forwards: 69
  280. Proxy-Authorization: Digest username="10022",realm="eu.sip.l7dev.co.cc",nonce="4dd6719b00000006748465b0050cdb908bf11e69caf3f4ad",uri="sip:501@eu.sip.l7dev.co.cc",response="6c2ee4856be83eda05301cc39e7b30ba",algorithm=MD5
  281. To: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  282. From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  283. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  284. CSeq: 638 ACK
  285. User-Agent: Twinkle/1.4.2
  286. Content-Length: 0
  287.  
  288. <------------->
  289. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 0 [ 37]: ACK sip:501@192.168.1.74:5060 SIP/2.0
  290. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bKf25f.87e67282.2
  291. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 2 [ 85]: Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKfbvogxma
  292. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69
  293. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 4 [219]: Proxy-Authorization: Digest username="10022",realm="eu.sip.l7dev.co.cc",nonce="4dd6719b00000006748465b0050cdb908bf11e69caf3f4ad",uri="sip:501@eu.sip.l7dev.co.cc",response="6c2ee4856be83eda05301cc39e7b30ba",algorithm=MD5
  294. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 5 [ 47]: To: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  295. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 6 [ 54]: From: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  296. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 7 [ 39]: Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  297. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 8 [ 13]: CSeq: 638 ACK
  298. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 9 [ 25]: User-Agent: Twinkle/1.4.2
  299. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 10 [ 17]: Content-Length: 0
  300. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: --- (11 headers 0 lines) ---
  301. [May 20 13:49:49] DEBUG[5717] chan_sip.c: = Looking for Call ID: ahmejjysyyetcgl@chris-ubuntu10 (Checking From) --From tag jdnch --To-tag as73bda3ac
  302. [May 20 13:49:49] DEBUG[5717] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  303. [May 20 13:49:49] DEBUG[5717] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6
  304. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Stopping retransmission on 'ahmejjysyyetcgl@chris-ubuntu10' of Response 638: Match Found
  305. [May 20 13:49:49] DEBUG[5719] res_rtp_asterisk.c: Got RTCP report of 40 bytes
  306. [May 20 13:49:49] DEBUG[5719] pbx.c: Launching 'ConfBridge'
  307. [May 20 13:49:49] VERBOSE[5719] pbx.c: -- Executing [501@common:3] ConfBridge("SIP/OpenSER-00000000", "1001") in new stack
  308. [May 20 13:49:49] DEBUG[5719] app_confbridge.c: Trying to find conference bridge '1001'
  309. [May 20 13:49:49] DEBUG[5719] app_confbridge.c: Destroying conference bridge '1001'
  310. [May 20 13:49:49] ERROR[5719] app_confbridge.c: Conference bridge '1001' could not be created.
  311. [May 20 13:49:49] DEBUG[5719] pbx.c: Spawn extension (common,501,3) exited non-zero on 'SIP/OpenSER-00000000'
  312. [May 20 13:49:49] VERBOSE[5719] pbx.c: == Spawn extension (common, 501, 3) exited non-zero on 'SIP/OpenSER-00000000'
  313. [May 20 13:49:49] DEBUG[5719] channel.c: Soft-Hanging up channel 'SIP/OpenSER-00000000'
  314. [May 20 13:49:49] DEBUG[5719] pbx.c: Launching 'Hangup'
  315. [May 20 13:49:49] VERBOSE[5719] pbx.c: -- Executing [h@common:1] Hangup("SIP/OpenSER-00000000", "") in new stack
  316. [May 20 13:49:49] DEBUG[5719] pbx.c: Spawn extension (common,h,1) exited non-zero on 'SIP/OpenSER-00000000'
  317. [May 20 13:49:49] VERBOSE[5719] pbx.c: == Spawn extension (common, h, 1) exited non-zero on 'SIP/OpenSER-00000000'
  318. [May 20 13:49:49] DEBUG[5719] channel.c: Hanging up channel 'SIP/OpenSER-00000000'
  319. [May 20 13:49:49] DEBUG[5719] chan_sip.c: Hangup call SIP/OpenSER-00000000, SIP callid ahmejjysyyetcgl@chris-ubuntu10
  320. [May 20 13:49:49] DEBUG[5719] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xdfdb98'
  321. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Scheduling destruction of SIP dialog 'ahmejjysyyetcgl@chris-ubuntu10' in 32000 ms (Method: ACK)
  322. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: set_destination: Parsing <sip:192.168.1.52;lr=on;ftag=jdnch> for address/port to send to
  323. [May 20 13:49:49] DEBUG[5719] netsock2.c: Splitting '192.168.1.52' gives...
  324. [May 20 13:49:49] DEBUG[5719] netsock2.c: ...host '192.168.1.52' and port '(null)'.
  325. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: set_destination: set destination to 192.168.1.52:5060
  326. [May 20 13:49:49] VERBOSE[5719] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.52:5060:
  327. BYE sip:10022@192.168.1.23:5060 SIP/2.0
  328.  
  329. Via: SIP/2.0/UDP 192.168.1.74:5060;branch=z9hG4bK4cfc7fe3;rport
  330.  
  331. Route: <sip:192.168.1.52;lr=on;ftag=jdnch>
  332.  
  333. Max-Forwards: 70
  334.  
  335. From: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  336.  
  337. To: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  338.  
  339. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  340.  
  341. CSeq: 102 BYE
  342.  
  343. User-Agent: Media GW 1
  344.  
  345. X-Asterisk-HangupCause: Normal Clearing
  346.  
  347. X-Asterisk-HangupCauseCode: 16
  348.  
  349. Content-Length: 0
  350.  
  351.  
  352.  
  353.  
  354. ---
  355. [May 20 13:49:49] DEBUG[5719] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9
  356. [May 20 13:49:49] DEBUG[5719] chan_sip.c: Trying to put 'BYE sip:100' onto UDP socket destined for 192.168.1.52:5060
  357. [May 20 13:49:49] DEBUG[5719] cdr.c: Dropping CDR !
  358. [May 20 13:49:49] DEBUG[5710] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  359. [May 20 13:49:49] DEBUG[5710] chan_sip.c: Checking device state for peer OpenSER
  360. [May 20 13:49:49] DEBUG[5710] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  361. [May 20 13:49:49] DEBUG[5710] devicestate.c: device 'SIP/OpenSER' state '1'
  362. [May 20 13:49:49] VERBOSE[5717] chan_sip.c:
  363. <--- SIP read from UDP:192.168.1.52:5060 --->
  364. SIP/2.0 200 OK
  365. Via: SIP/2.0/UDP 192.168.1.74:5060;received=192.168.1.74;rport=5060;branch=z9hG4bK4cfc7fe3
  366. To: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  367. From: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  368. Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  369. CSeq: 102 BYE
  370. Server: Twinkle/1.4.2
  371. Content-Length: 0
  372.  
  373. <------------->
  374. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  375. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.1.74:5060;received=192.168.1.74;rport=5060;branch=z9hG4bK4cfc7fe3
  376. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 2 [ 52]: To: "10022" <sip:10022@eu.sip.l7dev.co.cc>;tag=jdnch
  377. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 3 [ 49]: From: <sip:501@eu.sip.l7dev.co.cc>;tag=as73bda3ac
  378. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 4 [ 39]: Call-ID: ahmejjysyyetcgl@chris-ubuntu10
  379. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE
  380. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 6 [ 21]: Server: Twinkle/1.4.2
  381. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  382. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: --- (8 headers 0 lines) ---
  383. [May 20 13:49:49] DEBUG[5717] chan_sip.c: = Looking for Call ID: ahmejjysyyetcgl@chris-ubuntu10 (Checking To) --From tag as73bda3ac --To-tag jdnch
  384. [May 20 13:49:49] DEBUG[5717] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9
  385. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Stopping retransmission on 'ahmejjysyyetcgl@chris-ubuntu10' of Request 102: Match Found
  386. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
  387. [May 20 13:49:49] DEBUG[5717] chan_sip.c: Destroying SIP dialog ahmejjysyyetcgl@chris-ubuntu10
  388. [May 20 13:49:49] VERBOSE[5717] chan_sip.c: Really destroying SIP dialog 'ahmejjysyyetcgl@chris-ubuntu10' Method: ACK
  389. [May 20 13:49:49] DEBUG[5717] rtp_engine.c: Destroyed RTP instance '0xdfdb98'
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