API tools faq
paste
Advertisement
level7

console-1.8.4.log

level7
May 20th, 2011
146
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 24.04 KB | None | 0 0
raw download clone embed print report
  1. [May 20 13:47:02] VERBOSE[5140] chan_sip.c:
  2. <--- SIP read from UDP:192.168.1.52:5060 --->
  3. INVITE sip:[email protected] SIP/2.0
  4. Record-Route: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  5. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bK3061.f8641956.0
  6. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKnpxgmggd
  7. Max-Forwards: 69
  8. To: <sip:[email protected]>
  9. From: "10022" <sip:[email protected]>;tag=pmzgh
  10. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  11. CSeq: 929 INVITE
  12. Contact: <sip:[email protected]: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 1805319672 716775601 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 13:47:02] DEBUG[5140] chan_sip.c: Header 0 [ 41]: INVITE sip:[email protected] SIP/2.0
  35. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 1 [ 49]: Record-Route: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  36. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 2 [ 59]: Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bK3061.f8641956.0
  37. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 3 [ 85]: Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKnpxgmggd
  38. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 4 [ 16]: Max-Forwards: 69
  39. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 5 [ 32]: To: <sip:[email protected]>
  40. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 6 [ 54]: From: "10022" <sip:[email protected]>;tag=pmzgh
  41. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 7 [ 39]: Call-ID: eohiewieqwatxjz@chris-ubuntu10
  42. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 8 [ 16]: CSeq: 929 INVITE
  43. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 9 [ 38]: Contact: <sip:[email protected]:5060>
  44. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp
  45. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 11 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  46. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 12 [ 37]: Supported: replaces,norefersub,100rel
  47. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 13 [ 25]: User-Agent: Twinkle/1.4.2
  48. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 14 [ 19]: Content-Length: 308
  49. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 17 [ 0]:
  50. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 0 [ 3]: v=0
  51. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 1 [ 50]: o=twinkle 1805319672 716775601 IN IP4 192.168.1.23
  52. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 2 [ 3]: s=-
  53. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.23
  54. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 4 [ 5]: t=0 0
  55. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 5 [ 36]: m=audio 8000 RTP/AVP 98 97 8 0 3 101
  56. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 6 [ 23]: a=rtpmap:98 speex/16000
  57. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 7 [ 22]: a=rtpmap:97 speex/8000
  58. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000
  59. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000
  60. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000
  61. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000
  62. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15
  63. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Body 13 [ 10]: a=ptime:20
  64. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: --- (17 headers 14 lines) ---
  65. [May 20 13:47:02] DEBUG[5140] chan_sip.c: = Looking for Call ID: eohiewieqwatxjz@chris-ubuntu10 (Checking From) --From tag pmzgh --To-tag
  66. [May 20 13:47:02] DEBUG[5140] acl.c: For destination '192.168.1.52', our source address is '192.168.1.74'.
  67. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.74:5060
  68. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Allocating new SIP dialog for eohiewieqwatxjz@chris-ubuntu10 - INVITE (No RTP)
  69. [May 20 13:47:02] DEBUG[5140] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  70. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel"
  71. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Found SIP option: -replaces-
  72. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Matched SIP option: replaces
  73. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Found SIP option: -norefersub-
  74. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Matched SIP option: norefersub
  75. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Found SIP option: -100rel-
  76. [May 20 13:47:02] DEBUG[5140] sip/reqresp_parser.c: Matched SIP option: 100rel
  77. [May 20 13:47:02] DEBUG[5140] netsock2.c: Splitting '192.168.1.52' gives...
  78. [May 20 13:47:02] DEBUG[5140] netsock2.c: ...host '192.168.1.52' and port '(null)'.
  79. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Sending to 192.168.1.52:5060 (NAT)
  80. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Initializing initreq for method INVITE - callid eohiewieqwatxjz@chris-ubuntu10
  81. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Using INVITE request as basis request - eohiewieqwatxjz@chris-ubuntu10
  82. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found peer 'OpenSER' for '10022' from 192.168.1.52:5060
  83. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd93a38'
  84. [May 20 13:47:02] DEBUG[5140] res_rtp_asterisk.c: Allocated port 13548 for RTP instance '0xd93a38'
  85. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: RTP instance '0xd93a38' is setup and ready to go
  86. [May 20 13:47:02] DEBUG[5140] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd93a38'
  87. [May 20 13:47:02] VERBOSE[5140] netsock2.c: == Using SIP RTP CoS mark 5
  88. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Setting NAT on RTP to On
  89. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
  90. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing session-level SDP o=twinkle 1805319672 716775601 IN IP4 192.168.1.23... UNSUPPORTED.
  91. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED.
  92. [May 20 13:47:02] DEBUG[5140] netsock2.c: Splitting '192.168.1.23' gives...
  93. [May 20 13:47:02] DEBUG[5140] netsock2.c: ...host '192.168.1.23' and port '(null)'.
  94. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.23... OK.
  95. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
  96. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 98
  97. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 98 based on m type on 0x41faf440
  98. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 97
  99. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 97 based on m type on 0x41faf440
  100. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 8
  101. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 8 based on m type on 0x41faf440
  102. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 0
  103. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 0 based on m type on 0x41faf440
  104. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 3
  105. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 3 based on m type on 0x41faf440
  106. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found RTP audio format 101
  107. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Setting payload 101 based on m type on 0x41faf440
  108. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format speex for ID 98
  109. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 speex/16000... OK.
  110. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format speex for ID 97
  111. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 speex/8000... OK.
  112. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format PCMA for ID 8
  113. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
  114. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format PCMU for ID 0
  115. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  116. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format GSM for ID 3
  117. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
  118. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Found audio description format telephone-event for ID 101
  119. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  120. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
  121. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
  122. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 0 on 0x41faf440
  123. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 3 on 0x41faf440
  124. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 8 on 0x41faf440
  125. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 97 on 0x41faf440
  126. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 98 on 0x41faf440
  127. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Incorporating payload 101 on 0x41faf440
  128. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Capabilities: us - 0x1a0f (g723|gsm|ulaw|alaw|g726|speex|g722), peer - audio=0x20000020e (gsm|ulaw|alaw|speex|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20e (gsm|ulaw|alaw|speex)
  129. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  130. [May 20 13:47:02] DEBUG[5140] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd93a38'
  131. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Peer audio RTP is at port 192.168.1.23:8000
  132. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 0 from 0x41faf440 to 0xd93c00
  133. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 3 from 0x41faf440 to 0xd93c00
  134. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 8 from 0x41faf440 to 0xd93c00
  135. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 97 from 0x41faf440 to 0xd93c00
  136. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 98 from 0x41faf440 to 0xd93c00
  137. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Copying payload 101 from 0x41faf440 to 0xd93c00
  138. [May 20 13:47:02] DEBUG[5140] chan_sip.c: We're settling with these formats: 0x20e (gsm|ulaw|alaw|speex)
  139. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Checking SIP call limits for device
  140. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Updating call counter for incoming call
  141. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Looking for 501 in common (domain eu.sip.l7dev.co.cc)
  142. [May 20 13:47:02] DEBUG[5140] chan_sip.c: *** Our native formats are 0x8 (alaw)
  143. [May 20 13:47:02] DEBUG[5140] chan_sip.c: *** Joint capabilities are 0x20e (gsm|ulaw|alaw|speex)
  144. [May 20 13:47:02] DEBUG[5140] chan_sip.c: *** Our capabilities are 0x1a0f (g723|gsm|ulaw|alaw|g726|speex|g722)
  145. [May 20 13:47:02] DEBUG[5140] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
  146. [May 20 13:47:02] DEBUG[5140] chan_sip.c: This channel will not be able to handle video.
  147. [May 20 13:47:02] DEBUG[5140] chan_sip.c: build_route: Record-Route hop: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  148. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: list_route: hop: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  149. [May 20 13:47:02] DEBUG[5140] chan_sip.c: SIP/OpenSER-00000000: New call is still down.... Trying...
  150. [May 20 13:47:02] VERBOSE[5140] 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=z9hG4bK3061.f8641956.0;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=z9hG4bKnpxgmggd
  157.  
  158. Record-Route: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  159.  
  160. From: "10022" <sip:[email protected]>;tag=pmzgh
  161.  
  162. To: <sip:[email protected]>
  163.  
  164. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  165.  
  166. CSeq: 929 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:[email protected]:5060>
  175.  
  176. Content-Length: 0
  177.  
  178.  
  179.  
  180.  
  181. <------------>
  182. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.52:5060
  183. [May 20 13:47:02] DEBUG[5133] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  184. [May 20 13:47:02] DEBUG[5133] chan_sip.c: Checking device state for peer OpenSER
  185. [May 20 13:47:02] DEBUG[5133] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  186. [May 20 13:47:02] DEBUG[5133] devicestate.c: device 'SIP/OpenSER' state '1'
  187. [May 20 13:47:02] DEBUG[5191] pbx.c: Launching 'NoOp'
  188. [May 20 13:47:02] VERBOSE[5191] pbx.c: -- Executing [501@common:1] NoOp("SIP/OpenSER-00000000", "#### [common] ####") in new stack
  189. [May 20 13:47:02] DEBUG[5191] pbx.c: Launching 'Answer'
  190. [May 20 13:47:02] VERBOSE[5191] pbx.c: -- Executing [501@common:2] Answer("SIP/OpenSER-00000000", "") in new stack
  191. [May 20 13:47:02] DEBUG[5191] chan_sip.c: SIP answering channel: SIP/OpenSER-00000000
  192. [May 20 13:47:02] DEBUG[5191] res_rtp_asterisk.c: Setting the marker bit due to a source update
  193. [May 20 13:47:02] DEBUG[5191] chan_sip.c: Setting framing from config on incoming call
  194. [May 20 13:47:02] DEBUG[5191] chan_sip.c: ** Our capability: 0x20e (gsm|ulaw|alaw|speex) Video flag: True Text flag: True
  195. [May 20 13:47:02] DEBUG[5191] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  196. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Audio is at 5060
  197. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  198. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  199. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Adding codec 0x2 (gsm) to SDP
  200. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Adding codec 0x200 (speex) to SDP
  201. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  202. [May 20 13:47:02] DEBUG[5191] chan_sip.c: -- Done with adding codecs to SDP
  203. [May 20 13:47:02] DEBUG[5191] chan_sip.c: Done building SDP. Settling with this capability: 0x20e (gsm|ulaw|alaw|speex)
  204. [May 20 13:47:02] VERBOSE[5191] 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=z9hG4bK3061.f8641956.0;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=z9hG4bKnpxgmggd
  211.  
  212. Record-Route: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  213.  
  214. From: "10022" <sip:[email protected]>;tag=pmzgh
  215.  
  216. To: <sip:[email protected]>;tag=as50362fbf
  217.  
  218. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  219.  
  220. CSeq: 929 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:[email protected]:5060>
  229.  
  230. Content-Type: application/sdp
  231.  
  232. Content-Length: 325
  233.  
  234.  
  235.  
  236. v=0
  237.  
  238. o=root 624391397 624391397 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 13548 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 13:47:02] DEBUG[5191] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6
  269. [May 20 13:47:02] DEBUG[5191] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.52:5060
  270. [May 20 13:47:02] DEBUG[5133] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  271. [May 20 13:47:02] DEBUG[5133] chan_sip.c: Checking device state for peer OpenSER
  272. [May 20 13:47:02] DEBUG[5133] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  273. [May 20 13:47:02] DEBUG[5133] devicestate.c: device 'SIP/OpenSER' state '1'
  274. [May 20 13:47:02] VERBOSE[5140] chan_sip.c:
  275. <--- SIP read from UDP:192.168.1.52:5060 --->
  276. ACK sip:[email protected]:5060 SIP/2.0
  277. Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bK3061.f8641956.2
  278. Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKuhkbkmsj
  279. Max-Forwards: 69
  280. Proxy-Authorization: Digest username="10022",realm="eu.sip.l7dev.co.cc",nonce="4dd670f4000000055e12e55fb6f3b879e282269e9fe857bc",uri="sip:[email protected]",response="b77923632d37efd15c13a089e69d44b5",algorithm=MD5
  281. To: <sip:[email protected]>;tag=as50362fbf
  282. From: "10022" <sip:[email protected]>;tag=pmzgh
  283. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  284. CSeq: 929 ACK
  285. User-Agent: Twinkle/1.4.2
  286. Content-Length: 0
  287.  
  288. <------------->
  289. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 0 [ 37]: ACK sip:[email protected]:5060 SIP/2.0
  290. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.1.52;branch=z9hG4bK3061.f8641956.2
  291. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 2 [ 85]: Via: SIP/2.0/UDP 192.168.1.23;received=192.168.1.23;rport=5060;branch=z9hG4bKuhkbkmsj
  292. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69
  293. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 4 [219]: Proxy-Authorization: Digest username="10022",realm="eu.sip.l7dev.co.cc",nonce="4dd670f4000000055e12e55fb6f3b879e282269e9fe857bc",uri="sip:[email protected]",response="b77923632d37efd15c13a089e69d44b5",algorithm=MD5
  294. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 5 [ 47]: To: <sip:[email protected]>;tag=as50362fbf
  295. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 6 [ 54]: From: "10022" <sip:[email protected]>;tag=pmzgh
  296. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 7 [ 39]: Call-ID: eohiewieqwatxjz@chris-ubuntu10
  297. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 8 [ 13]: CSeq: 929 ACK
  298. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 9 [ 25]: User-Agent: Twinkle/1.4.2
  299. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 10 [ 17]: Content-Length: 0
  300. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: --- (11 headers 0 lines) ---
  301. [May 20 13:47:02] DEBUG[5140] chan_sip.c: = Looking for Call ID: eohiewieqwatxjz@chris-ubuntu10 (Checking From) --From tag pmzgh --To-tag as50362fbf
  302. [May 20 13:47:02] DEBUG[5140] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  303. [May 20 13:47:02] DEBUG[5140] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6
  304. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Stopping retransmission on 'eohiewieqwatxjz@chris-ubuntu10' of Response 929: Match Found
  305. [May 20 13:47:02] DEBUG[5191] res_rtp_asterisk.c: Got RTCP report of 40 bytes
  306. [May 20 13:47:02] DEBUG[5191] pbx.c: Launching 'ConfBridge'
  307. [May 20 13:47:02] VERBOSE[5191] pbx.c: -- Executing [501@common:3] ConfBridge("SIP/OpenSER-00000000", "1001") in new stack
  308. [May 20 13:47:02] DEBUG[5191] app_confbridge.c: Trying to find conference bridge '1001'
  309. [May 20 13:47:02] DEBUG[5191] bridging.c: Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)
  310. [May 20 13:47:02] DEBUG[5191] app_confbridge.c: Destroying conference bridge '1001'
  311. [May 20 13:47:02] ERROR[5191] app_confbridge.c: Conference bridge '1001' could not be created.
  312. [May 20 13:47:02] DEBUG[5191] pbx.c: Spawn extension (common,501,3) exited non-zero on 'SIP/OpenSER-00000000'
  313. [May 20 13:47:02] VERBOSE[5191] pbx.c: == Spawn extension (common, 501, 3) exited non-zero on 'SIP/OpenSER-00000000'
  314. [May 20 13:47:02] DEBUG[5191] channel.c: Soft-Hanging up channel 'SIP/OpenSER-00000000'
  315. [May 20 13:47:02] DEBUG[5191] pbx.c: Launching 'Hangup'
  316. [May 20 13:47:02] VERBOSE[5191] pbx.c: -- Executing [h@common:1] Hangup("SIP/OpenSER-00000000", "") in new stack
  317. [May 20 13:47:02] DEBUG[5191] pbx.c: Spawn extension (common,h,1) exited non-zero on 'SIP/OpenSER-00000000'
  318. [May 20 13:47:02] VERBOSE[5191] pbx.c: == Spawn extension (common, h, 1) exited non-zero on 'SIP/OpenSER-00000000'
  319. [May 20 13:47:02] DEBUG[5191] channel.c: Hanging up channel 'SIP/OpenSER-00000000'
  320. [May 20 13:47:02] DEBUG[5191] chan_sip.c: Hangup call SIP/OpenSER-00000000, SIP callid eohiewieqwatxjz@chris-ubuntu10
  321. [May 20 13:47:02] DEBUG[5191] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd93a38'
  322. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Scheduling destruction of SIP dialog 'eohiewieqwatxjz@chris-ubuntu10' in 32000 ms (Method: ACK)
  323. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: set_destination: Parsing <sip:192.168.1.52;lr=on;ftag=pmzgh> for address/port to send to
  324. [May 20 13:47:02] DEBUG[5191] netsock2.c: Splitting '192.168.1.52' gives...
  325. [May 20 13:47:02] DEBUG[5191] netsock2.c: ...host '192.168.1.52' and port '(null)'.
  326. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: set_destination: set destination to 192.168.1.52:5060
  327. [May 20 13:47:02] VERBOSE[5191] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.52:5060:
  328. BYE sip:[email protected]:5060 SIP/2.0
  329.  
  330. Via: SIP/2.0/UDP 192.168.1.74:5060;branch=z9hG4bK1ad11035;rport
  331.  
  332. Route: <sip:192.168.1.52;lr=on;ftag=pmzgh>
  333.  
  334. Max-Forwards: 70
  335.  
  336. From: <sip:[email protected]>;tag=as50362fbf
  337.  
  338. To: "10022" <sip:[email protected]>;tag=pmzgh
  339.  
  340. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  341.  
  342. CSeq: 102 BYE
  343.  
  344. User-Agent: Media GW 1
  345.  
  346. X-Asterisk-HangupCause: Normal Clearing
  347.  
  348. X-Asterisk-HangupCauseCode: 16
  349.  
  350. Content-Length: 0
  351.  
  352.  
  353.  
  354.  
  355. ---
  356. [May 20 13:47:02] DEBUG[5191] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #9
  357. [May 20 13:47:02] DEBUG[5191] chan_sip.c: Trying to put 'BYE sip:100' onto UDP socket destined for 192.168.1.52:5060
  358. [May 20 13:47:02] DEBUG[5191] cdr.c: Dropping CDR !
  359. [May 20 13:47:02] DEBUG[5133] devicestate.c: No provider found, checking channel drivers for SIP - OpenSER
  360. [May 20 13:47:02] DEBUG[5133] chan_sip.c: Checking device state for peer OpenSER
  361. [May 20 13:47:02] DEBUG[5133] devicestate.c: Changing state for SIP/OpenSER - state 1 (Not in use)
  362. [May 20 13:47:02] DEBUG[5133] devicestate.c: device 'SIP/OpenSER' state '1'
  363. [May 20 13:47:02] VERBOSE[5140] chan_sip.c:
  364. <--- SIP read from UDP:192.168.1.52:5060 --->
  365. SIP/2.0 200 OK
  366. Via: SIP/2.0/UDP 192.168.1.74:5060;received=192.168.1.74;rport=5060;branch=z9hG4bK1ad11035
  367. To: "10022" <sip:[email protected]>;tag=pmzgh
  368. From: <sip:[email protected]>;tag=as50362fbf
  369. Call-ID: eohiewieqwatxjz@chris-ubuntu10
  370. CSeq: 102 BYE
  371. Server: Twinkle/1.4.2
  372. Content-Length: 0
  373.  
  374. <------------->
  375. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
  376. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.1.74:5060;received=192.168.1.74;rport=5060;branch=z9hG4bK1ad11035
  377. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 2 [ 52]: To: "10022" <sip:[email protected]>;tag=pmzgh
  378. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 3 [ 49]: From: <sip:[email protected]>;tag=as50362fbf
  379. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 4 [ 39]: Call-ID: eohiewieqwatxjz@chris-ubuntu10
  380. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE
  381. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 6 [ 21]: Server: Twinkle/1.4.2
  382. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Header 7 [ 17]: Content-Length: 0
  383. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: --- (8 headers 0 lines) ---
  384. [May 20 13:47:02] DEBUG[5140] chan_sip.c: = Looking for Call ID: eohiewieqwatxjz@chris-ubuntu10 (Checking To) --From tag as50362fbf --To-tag pmzgh
  385. [May 20 13:47:02] DEBUG[5140] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9
  386. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Stopping retransmission on 'eohiewieqwatxjz@chris-ubuntu10' of Request 102: Match Found
  387. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
  388. [May 20 13:47:02] DEBUG[5140] chan_sip.c: Destroying SIP dialog eohiewieqwatxjz@chris-ubuntu10
  389. [May 20 13:47:02] VERBOSE[5140] chan_sip.c: Really destroying SIP dialog 'eohiewieqwatxjz@chris-ubuntu10' Method: ACK
  390. [May 20 13:47:02] DEBUG[5140] rtp_engine.c: Destroyed RTP instance '0xd93a38'
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!