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  1. *CLI>
  2. <--- SIP read from UDP://78.105.1.131:5061 --->
  3. INVITE sip:[email protected] SIP/2.0
  4. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKwrwgiguw
  5. Max-Forwards: 70
  6. To: <sip:[email protected]>
  7. From: "10000" <sip:[email protected]>;tag=cywki
  8. Call-ID: [email protected]
  9. CSeq: 960 INVITE
  10. Contact: <sip:[email protected]:5061>
  11. Content-Type: application/sdp
  12. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  13. Supported: replaces,norefersub,100rel
  14. User-Agent: Twinkle/1.1
  15. Content-Length: 190
  16.  
  17. v=0
  18. o=10000 546776892 434197002 IN IP4 192.168.7.55
  19. s=-
  20. c=IN IP4 78.105.1.131
  21. t=0 0
  22. m=audio 8002 RTP/AVP 3 101
  23. a=rtpmap:3 GSM/8000
  24. a=rtpmap:101 telephone-event/8000
  25. a=fmtp:101 0-15
  26.  
  27. <------------->
  28. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 0 [ 41]: INVITE sip:[email protected] SIP/2.0
  29. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKwrwgiguw
  30. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  31. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 3 [ 32]: To: <sip:[email protected]>
  32. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=cywki
  33. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  34. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 6 [ 16]: CSeq: 960 INVITE
  35. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 7 [ 38]: Contact: <sip:[email protected]:5061>
  36. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 8 [ 29]: Content-Type: application/sdp
  37. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  38. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 10 [ 37]: Supported: replaces,norefersub,100rel
  39. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 11 [ 23]: User-Agent: Twinkle/1.1
  40. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 12 [ 19]: Content-Length: 190
  41. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 13 [ 0]:
  42. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 0 [ 3]: v=0
  43. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 1 [ 47]: o=10000 546776892 434197002 IN IP4 192.168.7.55
  44. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 2 [ 3]: s=-
  45. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 3 [ 21]: c=IN IP4 78.105.1.131
  46. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 4 [ 5]: t=0 0
  47. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 5 [ 26]: m=audio 8002 RTP/AVP 3 101
  48. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 6 [ 19]: a=rtpmap:3 GSM/8000
  49. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  50. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 8 [ 15]: a=fmtp:101 0-15
  51. --- (13 headers 9 lines) ---
  52. [May 21 09:31:48] DEBUG[6835]: acl.c:490 ast_ouraddrfor: Found IP address for this socket
  53. == Using SIP RTP CoS mark 5
  54. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off
  55. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6350 sip_alloc: Allocating new SIP dialog for [email protected] - INVITE (With RTP)
  56. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:19645 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  57. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2742 parse_sip_options: Begin: parsing SIP "Supported: replaces,norefersub,100rel"
  58. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2750 parse_sip_options: Found SIP option: -replaces-
  59. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2756 parse_sip_options: Matched SIP option: replaces
  60. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2750 parse_sip_options: Found SIP option: -norefersub-
  61. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2756 parse_sip_options: Matched SIP option: norefersub
  62. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2750 parse_sip_options: Found SIP option: -100rel-
  63. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2756 parse_sip_options: Matched SIP option: 100rel
  64. Sending to 78.105.1.131 : 5061 (NAT)
  65. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:18059 handle_request_invite: Initializing initreq for method INVITE - callid [email protected]
  66. Using INVITE request as basis request - [email protected]
  67. Found peer '10000' for '10000' from 78.105.1.131:5061
  68. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to On
  69.  
  70. <--- Reliably Transmitting (NAT) to 78.105.1.131:5061 --->
  71. SIP/2.0 401 Unauthorized
  72. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKwrwgiguw;received=78.105.1.131;rport=5061
  73. From: "10000" <sip:[email protected]>;tag=cywki
  74. To: <sip:[email protected]>;tag=as6e1023a9
  75. Call-ID: [email protected]
  76. CSeq: 960 INVITE
  77. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  78. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  79. Supported: replaces, timer
  80. WWW-Authenticate: Digest algorithm=MD5, realm="dev-sip.wima.co.uk", nonce="4f0ecb73"
  81. Content-Length: 0
  82.  
  83.  
  84. <------------>
  85. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3200 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #7
  86. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 40' onto UDP socket destined for 78.105.1.131:5061
  87. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  88.  
  89. <--- SIP read from UDP://78.105.1.131:5061 --->
  90. ACK sip:[email protected] SIP/2.0
  91. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKwrwgiguw
  92. Max-Forwards: 70
  93. To: <sip:[email protected]>;tag=as6e1023a9
  94. From: "10000" <sip:[email protected]>;tag=cywki
  95. Call-ID: [email protected]
  96. CSeq: 960 ACK
  97. User-Agent: Twinkle/1.1
  98. Content-Length: 0
  99.  
  100.  
  101. <------------->
  102. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 0 [ 38]: ACK sip:[email protected] SIP/2.0
  103. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKwrwgiguw
  104. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  105. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 3 [ 47]: To: <sip:[email protected]>;tag=as6e1023a9
  106. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=cywki
  107. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  108. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 6 [ 13]: CSeq: 960 ACK
  109. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 7 [ 23]: User-Agent: Twinkle/1.1
  110. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 8 [ 17]: Content-Length: 0
  111. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 9 [ 0]:
  112. --- (9 headers 0 lines) ---
  113. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:19645 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  114. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3339 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7
  115. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3371 __sip_ack: Stopping retransmission on '[email protected]' of Response 960: Match Found
  116.  
  117. <--- SIP read from UDP://78.105.1.131:5061 --->
  118. INVITE sip:[email protected] SIP/2.0
  119. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKfssddmlh
  120. Max-Forwards: 70
  121. To: <sip:[email protected]>
  122. From: "10000" <sip:[email protected]>;tag=cywki
  123. Call-ID: [email protected]
  124. CSeq: 961 INVITE
  125. Contact: <sip:[email protected]:5061>
  126. Content-Type: application/sdp
  127. Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="4f0ecb73",uri="sip:[email protected]",response="f8bcb1d21bf34bb5bef9447dcf9bf8a3",algorithm=MD5
  128. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  129. Supported: replaces,norefersub,100rel
  130. User-Agent: Twinkle/1.1
  131. Content-Length: 190
  132.  
  133. v=0
  134. o=10000 546776892 434197002 IN IP4 192.168.7.55
  135. s=-
  136. c=IN IP4 78.105.1.131
  137. t=0 0
  138. m=audio 8002 RTP/AVP 3 101
  139. a=rtpmap:3 GSM/8000
  140. a=rtpmap:101 telephone-event/8000
  141. a=fmtp:101 0-15
  142.  
  143. <------------->
  144. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 0 [ 41]: INVITE sip:[email protected] SIP/2.0
  145. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKfssddmlh
  146. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  147. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 3 [ 32]: To: <sip:[email protected]>
  148. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=cywki
  149. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  150. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 6 [ 16]: CSeq: 961 INVITE
  151. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 7 [ 38]: Contact: <sip:[email protected]:5061>
  152. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 8 [ 29]: Content-Type: application/sdp
  153. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 9 [173]: Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="4f0ecb73",uri="sip:[email protected]",response="f8bcb1d21bf34bb5bef9447dcf9bf8a3",algorithm=MD5
  154. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 10 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  155. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 11 [ 37]: Supported: replaces,norefersub,100rel
  156. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 12 [ 23]: User-Agent: Twinkle/1.1
  157. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 13 [ 19]: Content-Length: 190
  158. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 14 [ 0]:
  159. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 0 [ 3]: v=0
  160. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 1 [ 47]: o=10000 546776892 434197002 IN IP4 192.168.7.55
  161. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 2 [ 3]: s=-
  162. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 3 [ 21]: c=IN IP4 78.105.1.131
  163. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 4 [ 5]: t=0 0
  164. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 5 [ 26]: m=audio 8002 RTP/AVP 3 101
  165. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 6 [ 19]: a=rtpmap:3 GSM/8000
  166. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  167. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Body 8 [ 15]: a=fmtp:101 0-15
  168. --- (14 headers 9 lines) ---
  169. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:19645 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  170. Sending to 78.105.1.131 : 5061 (NAT)
  171. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:18059 handle_request_invite: Initializing initreq for method INVITE - callid [email protected]
  172. Using INVITE request as basis request - [email protected]
  173. Found peer '10000' for '10000' from 78.105.1.131:5061
  174. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to On
  175. Found RTP audio format 3
  176. Found RTP audio format 101
  177. Peer audio RTP is at port 78.105.1.131:8002
  178. Found audio description format GSM for ID 3
  179. Found audio description format telephone-event for ID 101
  180. Got unsupported a:fmtp in SDP offer
  181. Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  182. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  183. Peer audio RTP is at port 78.105.1.131:8002
  184. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:7558 process_sdp: We're settling with these formats: 0x2 (gsm)
  185. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:18136 handle_request_invite: Checking SIP call limits for device 10000
  186. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:4815 update_call_counter: Updating call counter for incoming call
  187. Looking for 501 in common (domain dev-sip.wima.co.uk)
  188. [May 21 09:31:48] DEBUG[6835]: pbx.c:1357 new_find_extension: Nothing strange about this match
  189. [May 21 09:31:48] DEBUG[6835]: pbx.c:1357 new_find_extension: Nothing strange about this match
  190. [May 21 09:31:48] DEBUG[6835]: pbx.c:1357 new_find_extension: Nothing strange about this match
  191. [May 21 09:31:48] DEBUG[6835]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  192. [May 21 09:31:48] DEBUG[6835]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  193. [May 21 09:31:48] DEBUG[6835]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  194. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:5775 sip_new: *** Our native formats are 0x2 (gsm)
  195. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:5776 sip_new: *** Joint capabilities are 0x2 (gsm)
  196. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:5777 sip_new: *** Our capabilities are 0x100f (g723|gsm|ulaw|alaw|g722)
  197. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:5778 sip_new: *** AST_CODEC_CHOOSE formats are 0x2 (gsm)
  198. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:5808 sip_new: This channel will not be able to handle video.
  199. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:10853 build_route: build_route: Contact hop: <sip:[email protected]:5061>
  200. list_route: hop: <sip:[email protected]:5061>
  201. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:18349 handle_request_invite: SIP/10000-08226208: New call is still down.... Trying...
  202.  
  203. <--- Transmitting (NAT) to 78.105.1.131:5061 --->
  204. SIP/2.0 100 Trying
  205. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKfssddmlh;received=78.105.1.131;rport=5061
  206. From: "10000" <sip:[email protected]>;tag=cywki
  207. To: <sip:[email protected]>
  208. Call-ID: [email protected]
  209. CSeq: 961 INVITE
  210. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  211. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  212. Supported: replaces, timer
  213. Contact: <sip:[email protected]>
  214. Content-Length: 0
  215.  
  216.  
  217. <------------>
  218. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 10' onto UDP socket destined for 78.105.1.131:5061
  219. [May 21 09:31:48] DEBUG[6832]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  220. [May 21 09:31:48] DEBUG[6832]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  221. [May 21 09:31:48] DEBUG[6832]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  222. [May 21 09:31:48] DEBUG[6832]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  223. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  224. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  225. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  226. [May 21 09:31:48] DEBUG[6839]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  227. [May 21 09:31:48] DEBUG[6839]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  228. [May 21 09:31:48] DEBUG[6839]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  229. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  230. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  231. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  232. [May 21 09:31:48] DEBUG[6839]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  233. [May 21 09:31:48] DEBUG[6839]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  234. [May 21 09:31:48] DEBUG[6839]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  235. [May 21 09:31:48] DEBUG[6839]: pbx.c:3179 pbx_extension_helper: Launching 'MeetMe'
  236. -- Executing [501@common:1] MeetMe("SIP/10000-08226208", "12,MI") in new stack
  237. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:5347 sip_answer: SIP answering channel: SIP/10000-08226208
  238. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:9013 transmit_response_with_sdp: Setting framing from config on incoming call
  239. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:8681 add_sdp: ** Our capability: 0x2 (gsm) Video flag: True Text flag: True
  240. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:8682 add_sdp: ** Our prefcodec: 0x0 (nothing)
  241. Audio is at 78.105.1.127 port 15762
  242. Adding codec 0x2 (gsm) to SDP
  243. Adding non-codec 0x1 (telephone-event) to SDP
  244. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:8830 add_sdp: -- Done with adding codecs to SDP
  245. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:8949 add_sdp: Done building SDP. Settling with this capability: 0x2 (gsm)
  246.  
  247. <--- Reliably Transmitting (NAT) to 78.105.1.131:5061 --->
  248. SIP/2.0 200 OK
  249. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKfssddmlh;received=78.105.1.131;rport=5061
  250. From: "10000" <sip:[email protected]>;tag=cywki
  251. To: <sip:[email protected]>;tag=as6d6e102c
  252. Call-ID: [email protected]
  253. CSeq: 961 INVITE
  254. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  255. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  256. Supported: replaces, timer
  257. Contact: <sip:[email protected]>
  258. Content-Type: application/sdp
  259. Content-Length: 280
  260.  
  261. v=0
  262. o=root 1540854398 1540854398 IN IP4 78.105.1.127
  263. s=Asterisk PBX SVN-branch-1.6.1-r190371M
  264. c=IN IP4 78.105.1.127
  265. t=0 0
  266. m=audio 15762 RTP/AVP 3 101
  267. a=rtpmap:3 GSM/8000
  268. a=rtpmap:101 telephone-event/8000
  269. a=fmtp:101 0-16
  270. a=silenceSupp:off - - - -
  271. a=ptime:20
  272. a=sendrecv
  273.  
  274. <------------>
  275. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:3200 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #9
  276. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 78.105.1.131:5061
  277. [May 21 09:31:48] DEBUG[6832]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  278. [May 21 09:31:48] DEBUG[6832]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  279. [May 21 09:31:48] DEBUG[6832]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  280. [May 21 09:31:48] DEBUG[6832]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  281.  
  282. <--- SIP read from UDP://78.105.1.131:5061 --->
  283. ACK sip:[email protected] SIP/2.0
  284. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKuuqqloyw
  285. Max-Forwards: 70
  286. To: <sip:[email protected]>;tag=as6d6e102c
  287. From: "10000" <sip:[email protected]>;tag=cywki
  288. Call-ID: [email protected]
  289. CSeq: 961 ACK
  290. Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="4f0ecb73",uri="sip:[email protected]",response="f8bcb1d21bf34bb5bef9447dcf9bf8a3",algorithm=MD5
  291. User-Agent: Twinkle/1.1
  292. Content-Length: 0
  293.  
  294.  
  295. <------------->
  296. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 0 [ 32]: ACK sip:[email protected] SIP/2.0
  297. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKuuqqloyw
  298. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  299. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 3 [ 47]: To: <sip:[email protected]>;tag=as6d6e102c
  300. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=cywki
  301. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  302. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 6 [ 13]: CSeq: 961 ACK
  303. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 7 [173]: Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="4f0ecb73",uri="sip:[email protected]",response="f8bcb1d21bf34bb5bef9447dcf9bf8a3",algorithm=MD5
  304. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 8 [ 23]: User-Agent: Twinkle/1.1
  305. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 9 [ 17]: Content-Length: 0
  306. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 10 [ 0]:
  307. --- (10 headers 0 lines) ---
  308. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:19645 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  309. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3339 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9
  310. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3371 __sip_ack: Stopping retransmission on '[email protected]' of Response 961: Match Found
  311. [May 21 09:31:48] DEBUG[6839]: channel.c:1764 __ast_answer: Didn't receive a media frame from SIP/10000-08226208 within 500 ms of answering. Continuing anyway
  312. [May 21 09:31:48] DEBUG[6839]: app_meetme.c:3030 find_conf: The requested confno is '12'?
  313. == Parsing '/etc/asterisk/meetme.conf': [May 21 09:31:48] DEBUG[6839]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf
  314. == Found
  315. [May 21 09:31:48] DEBUG[6839]: app_meetme.c:3082 find_conf: 12 isn't a valid conference
  316. [May 21 09:31:48] DEBUG[6839]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay.
  317. [May 21 09:31:48] DEBUG[6839]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12'
  318. [May 21 09:31:48] DEBUG[6839]: chan_dahdi.c:9549 dahdi_request: Using channel -2
  319. [May 21 09:31:48] DEBUG[6839]: devicestate.c:467 devstate_event: device 'DAHDI/pseudo-1265532284' state '2'
  320. @@[May 21 09:31:48] DEBUG[6839]: channel.c:3475 set_format: Set channel DAHDI/pseudo-1265532284 to read format slin
  321. @@[May 21 09:31:48] DEBUG[6839]: channel.c:3475 set_format: Set channel DAHDI/pseudo-1265532284 to write format slin
  322. @@ -- Created MeetMe conference 1023 for conference '12'
  323. @@[May 21 09:31:48] DEBUG[6839]: app.c:645 __ast_play_and_record: play_and_record: vm-rec-name, /var/spool/asterisk/meetme/meetme-username-12-1, 'sln'
  324. @@[May 21 09:31:48] DEBUG[6839]: channel.c:3475 set_format: Set channel SIP/10000-08226208 to write format slin
  325. @@ -- <SIP/10000-08226208> Playing 'vm-rec-name.slin' (language 'en')
  326. @@[May 21 09:31:48] DEBUG[6839]: devicestate.c:467 devstate_event: device 'meetme:12' state '1'
  327. @@[May 21 09:31:48] DEBUG[6839]: channel.c:1549 ast_softhangup_nolock: Soft-Hanging up channel 'DAHDI/pseudo-1265532284'
  328. @@[May 21 09:31:48] DEBUG[6839]: channel.c:1644 ast_hangup: Hanging up channel 'DAHDI/pseudo-1265532284'
  329. @@[May 21 09:31:48] DEBUG[6839]: chan_dahdi.c:3381 dahdi_hangup: dahdi_hangup(DAHDI/pseudo-1265532284)
  330. @@[May 21 09:31:48] DEBUG[6839]: chan_dahdi.c:3415 dahdi_hangup: Hangup: channel: -2 index = 0, normal = 24, callwait = -1, thirdcall = -1
  331. [May 21 09:31:48] DEBUG[6839]: chan_dahdi.c:3888 dahdi_setoption: Set option TDD MODE, value: OFF(0) on DAHDI/pseudo-1265532284
  332. [May 21 09:31:48] DEBUG[6839]: chan_dahdi.c:2000 update_conf: Updated conferencing on -2, with 0 conference users
  333. -- Hungup 'DAHDI/pseudo-1265532284'
  334. [May 21 09:31:48] DEBUG[6832]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for DAHDI - pseudo
  335. [May 21 09:31:48] DEBUG[6832]: devicestate.c:487 do_state_change: Changing state for DAHDI/pseudo - state 0 (Unknown)
  336. [May 21 09:31:48] DEBUG[6832]: devicestate.c:467 devstate_event: device 'DAHDI/pseudo' state '0'
  337. [May 21 09:31:48] DEBUG[6839]: pbx.c:3779 __ast_pbx_run: Spawn extension (common,501,1) exited non-zero on 'SIP/10000-08226208'
  338. == Spawn extension (common, 501, 1) exited non-zero on 'SIP/10000-08226208'
  339. [May 21 09:31:48] DEBUG[6839]: channel.c:1549 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/10000-08226208'
  340. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  341. [May 21 09:31:48] DEBUG[6839]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- h
  342. [May 21 09:31:48] DEBUG[6839]: pbx.c:1357 new_find_extension: Nothing strange about this match
  343. [May 21 09:31:48] DEBUG[6839]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- h
  344. [May 21 09:31:48] DEBUG[6839]: pbx.c:3179 pbx_extension_helper: Launching 'Hangup'
  345. -- Executing [h@common:1] Hangup("SIP/10000-08226208", "") in new stack
  346. [May 21 09:31:48] DEBUG[6839]: pbx.c:3922 __ast_pbx_run: Spawn extension (common,h,1) exited non-zero on 'SIP/10000-08226208'
  347. == Spawn extension (common, h, 1) exited non-zero on 'SIP/10000-08226208'
  348. [May 21 09:31:48] DEBUG[6839]: channel.c:1644 ast_hangup: Hanging up channel 'SIP/10000-08226208'
  349. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:5176 sip_hangup: Hangup call SIP/10000-08226208, SIP callid [email protected]
  350. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: ACK)
  351. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:8065 reqprep: Strict routing enforced for session [email protected]
  352. set_destination: Parsing <sip:[email protected]:5061> for address/port to send to
  353. set_destination: set destination to 78.105.1.131, port 5061
  354. Reliably Transmitting (NAT) to 78.105.1.131:5061:
  355. BYE sip:[email protected]:5061 SIP/2.0
  356. Via: SIP/2.0/UDP 78.105.1.127:5060;branch=z9hG4bK1b2b2989;rport
  357. Max-Forwards: 70
  358. From: <sip:[email protected]>;tag=as6d6e102c
  359. To: "10000" <sip:[email protected]>;tag=cywki
  360. Call-ID: [email protected]
  361. CSeq: 102 BYE
  362. User-Agent: Asterisk PBX SVN-branch-1.6.1-r190371M
  363. X-Asterisk-HangupCause: Normal Clearing
  364. X-Asterisk-HangupCauseCode: 16
  365. Content-Length: 0
  366.  
  367.  
  368. ---
  369. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:3200 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #11
  370. [May 21 09:31:48] DEBUG[6839]: chan_sip.c:2870 __sip_xmit: Trying to put 'BYE sip:10' onto UDP socket destined for 78.105.1.131:5061
  371. [May 21 09:31:48] DEBUG[6839]: cdr.c:1209 ast_cdr_detach: Dropping CDR !
  372. [May 21 09:31:48] DEBUG[6832]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  373. [May 21 09:31:48] DEBUG[6832]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  374. [May 21 09:31:48] DEBUG[6832]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  375. [May 21 09:31:48] DEBUG[6832]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  376.  
  377. <--- SIP read from UDP://78.105.1.131:5061 --->
  378. SIP/2.0 200 OK
  379. Via: SIP/2.0/UDP 78.105.1.127:5060;rport=5060;branch=z9hG4bK1b2b2989
  380. To: "10000" <sip:[email protected]>;tag=cywki
  381. From: <sip:[email protected]>;tag=as6d6e102c
  382. Call-ID: [email protected]
  383. CSeq: 102 BYE
  384. Server: Twinkle/1.1
  385. Content-Length: 0
  386.  
  387.  
  388. <------------->
  389. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 0 [ 14]: SIP/2.0 200 OK
  390. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 1 [ 68]: Via: SIP/2.0/UDP 78.105.1.127:5060;rport=5060;branch=z9hG4bK1b2b2989
  391. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 2 [ 52]: To: "10000" <sip:[email protected]>;tag=cywki
  392. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 3 [ 49]: From: <sip:[email protected]>;tag=as6d6e102c
  393. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 4 [ 37]: Call-ID: [email protected]
  394. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 5 [ 13]: CSeq: 102 BYE
  395. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 6 [ 19]: Server: Twinkle/1.1
  396. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 7 [ 17]: Content-Length: 0
  397. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:6691 parse_request: Header 8 [ 0]:
  398. --- (8 headers 0 lines) ---
  399. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3339 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11
  400. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:3371 __sip_ack: Stopping retransmission on '[email protected]' of Request 102: Match Found
  401. SIP Response message for INCOMING dialog BYE arrived
  402. [May 21 09:31:48] DEBUG[6835]: chan_sip.c:4962 sip_destroy: Destroying SIP dialog [email protected]
  403. Really destroying SIP dialog '[email protected]' Method: ACK
  404.  
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