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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=z9hG4bKgzfpxncr
  5. Max-Forwards: 70
  6. To: <sip:[email protected]>
  7. From: "10000" <sip:[email protected]>;tag=njogt
  8. Call-ID: [email protected]
  9. CSeq: 865 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: 204
  16.  
  17. v=0
  18. o=10000 631264428 2046292071 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 8 101
  23. a=rtpmap:8 PCMA/8000
  24. a=rtpmap:101 telephone-event/8000
  25. a=fmtp:101 0-15
  26. a=ptime:20
  27.  
  28. <------------->
  29. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 0 [ 41]: INVITE sip:[email protected] SIP/2.0
  30. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKgzfpxncr
  31. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  32. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 3 [ 32]: To: <sip:[email protected]>
  33. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=njogt
  34. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  35. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 6 [ 16]: CSeq: 865 INVITE
  36. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 7 [ 38]: Contact: <sip:[email protected]:5061>
  37. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 8 [ 29]: Content-Type: application/sdp
  38. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  39. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 10 [ 37]: Supported: replaces,norefersub,100rel
  40. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 11 [ 23]: User-Agent: Twinkle/1.1
  41. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 12 [ 19]: Content-Length: 204
  42. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 13 [ 0]:
  43. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 0 [ 3]: v=0
  44. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 1 [ 48]: o=10000 631264428 2046292071 IN IP4 192.168.7.55
  45. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 2 [ 3]: s=-
  46. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 3 [ 21]: c=IN IP4 78.105.1.131
  47. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 4 [ 5]: t=0 0
  48. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 5 [ 26]: m=audio 8002 RTP/AVP 8 101
  49. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000
  50. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  51. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 8 [ 15]: a=fmtp:101 0-15
  52. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 9 [ 10]: a=ptime:20
  53. --- (13 headers 10 lines) ---
  54. [May 21 09:33:22] DEBUG[6868]: acl.c:490 ast_ouraddrfor: Found IP address for this socket
  55. == Using SIP RTP CoS mark 5
  56. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off
  57. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6350 sip_alloc: Allocating new SIP dialog for [email protected] - INVITE (With RTP)
  58. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:19645 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  59. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2742 parse_sip_options: Begin: parsing SIP "Supported: replaces,norefersub,100rel"
  60. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2750 parse_sip_options: Found SIP option: -replaces-
  61. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2756 parse_sip_options: Matched SIP option: replaces
  62. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2750 parse_sip_options: Found SIP option: -norefersub-
  63. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2756 parse_sip_options: Matched SIP option: norefersub
  64. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2750 parse_sip_options: Found SIP option: -100rel-
  65. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2756 parse_sip_options: Matched SIP option: 100rel
  66. Sending to 78.105.1.131 : 5061 (NAT)
  67. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:18059 handle_request_invite: Initializing initreq for method INVITE - callid [email protected]
  68. Using INVITE request as basis request - [email protected]
  69. Found peer '10000' for '10000' from 78.105.1.131:5061
  70. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to On
  71.  
  72. <--- Reliably Transmitting (NAT) to 78.105.1.131:5061 --->
  73. SIP/2.0 401 Unauthorized
  74. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKgzfpxncr;received=78.105.1.131;rport=5061
  75. From: "10000" <sip:[email protected]>;tag=njogt
  76. To: <sip:[email protected]>;tag=as1363d1c3
  77. Call-ID: [email protected]
  78. CSeq: 865 INVITE
  79. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  80. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  81. Supported: replaces, timer
  82. WWW-Authenticate: Digest algorithm=MD5, realm="dev-sip.wima.co.uk", nonce="7f72e534"
  83. Content-Length: 0
  84.  
  85.  
  86. <------------>
  87. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:3200 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #7
  88. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 40' onto UDP socket destined for 78.105.1.131:5061
  89. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  90.  
  91. <--- SIP read from UDP://78.105.1.131:5061 --->
  92. ACK sip:[email protected] SIP/2.0
  93. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKgzfpxncr
  94. Max-Forwards: 70
  95. To: <sip:[email protected]>;tag=as1363d1c3
  96. From: "10000" <sip:[email protected]>;tag=njogt
  97. Call-ID: [email protected]
  98. CSeq: 865 ACK
  99. User-Agent: Twinkle/1.1
  100. Content-Length: 0
  101.  
  102.  
  103. <------------->
  104. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 0 [ 38]: ACK sip:[email protected] SIP/2.0
  105. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKgzfpxncr
  106. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  107. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 3 [ 47]: To: <sip:[email protected]>;tag=as1363d1c3
  108. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=njogt
  109. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  110. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 6 [ 13]: CSeq: 865 ACK
  111. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 7 [ 23]: User-Agent: Twinkle/1.1
  112. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 8 [ 17]: Content-Length: 0
  113. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 9 [ 0]:
  114. --- (9 headers 0 lines) ---
  115. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:19645 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  116. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:3339 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7
  117. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:3371 __sip_ack: Stopping retransmission on '[email protected]' of Response 865: Match Found
  118.  
  119. <--- SIP read from UDP://78.105.1.131:5061 --->
  120. INVITE sip:[email protected] SIP/2.0
  121. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKyinmzjzo
  122. Max-Forwards: 70
  123. To: <sip:[email protected]>
  124. From: "10000" <sip:[email protected]>;tag=njogt
  125. Call-ID: [email protected]
  126. CSeq: 866 INVITE
  127. Contact: <sip:[email protected]:5061>
  128. Content-Type: application/sdp
  129. Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="7f72e534",uri="sip:[email protected]",response="eae8c2ab46c4ef7f9dc1d97bf2f58ef7",algorithm=MD5
  130. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  131. Supported: replaces,norefersub,100rel
  132. User-Agent: Twinkle/1.1
  133. Content-Length: 204
  134.  
  135. v=0
  136. o=10000 631264428 2046292071 IN IP4 192.168.7.55
  137. s=-
  138. c=IN IP4 78.105.1.131
  139. t=0 0
  140. m=audio 8002 RTP/AVP 8 101
  141. a=rtpmap:8 PCMA/8000
  142. a=rtpmap:101 telephone-event/8000
  143. a=fmtp:101 0-15
  144. a=ptime:20
  145.  
  146. <------------->
  147. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 0 [ 41]: INVITE sip:[email protected] SIP/2.0
  148. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKyinmzjzo
  149. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  150. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 3 [ 32]: To: <sip:[email protected]>
  151. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=njogt
  152. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  153. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 6 [ 16]: CSeq: 866 INVITE
  154. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 7 [ 38]: Contact: <sip:[email protected]:5061>
  155. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 8 [ 29]: Content-Type: application/sdp
  156. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 9 [173]: Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="7f72e534",uri="sip:[email protected]",response="eae8c2ab46c4ef7f9dc1d97bf2f58ef7",algorithm=MD5
  157. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 10 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  158. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 11 [ 37]: Supported: replaces,norefersub,100rel
  159. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 12 [ 23]: User-Agent: Twinkle/1.1
  160. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 13 [ 19]: Content-Length: 204
  161. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 14 [ 0]:
  162. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 0 [ 3]: v=0
  163. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 1 [ 48]: o=10000 631264428 2046292071 IN IP4 192.168.7.55
  164. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 2 [ 3]: s=-
  165. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 3 [ 21]: c=IN IP4 78.105.1.131
  166. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 4 [ 5]: t=0 0
  167. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 5 [ 26]: m=audio 8002 RTP/AVP 8 101
  168. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 6 [ 20]: a=rtpmap:8 PCMA/8000
  169. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
  170. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 8 [ 15]: a=fmtp:101 0-15
  171. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Body 9 [ 10]: a=ptime:20
  172. --- (14 headers 10 lines) ---
  173. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:19645 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  174. Sending to 78.105.1.131 : 5061 (NAT)
  175. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:18059 handle_request_invite: Initializing initreq for method INVITE - callid [email protected]
  176. Using INVITE request as basis request - [email protected]
  177. Found peer '10000' for '10000' from 78.105.1.131:5061
  178. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to On
  179. Found RTP audio format 8
  180. Found RTP audio format 101
  181. Peer audio RTP is at port 78.105.1.131:8002
  182. Found audio description format PCMA for ID 8
  183. Found audio description format telephone-event for ID 101
  184. Got unsupported a:fmtp in SDP offer
  185. Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  186. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  187. Peer audio RTP is at port 78.105.1.131:8002
  188. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:7558 process_sdp: We're settling with these formats: 0x8 (alaw)
  189. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:18136 handle_request_invite: Checking SIP call limits for device 10000
  190. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:4815 update_call_counter: Updating call counter for incoming call
  191. Looking for 501 in common (domain dev-sip.wima.co.uk)
  192. [May 21 09:33:22] DEBUG[6868]: pbx.c:1357 new_find_extension: Nothing strange about this match
  193. [May 21 09:33:22] DEBUG[6868]: pbx.c:1357 new_find_extension: Nothing strange about this match
  194. [May 21 09:33:22] DEBUG[6868]: pbx.c:1357 new_find_extension: Nothing strange about this match
  195. [May 21 09:33:22] DEBUG[6868]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  196. [May 21 09:33:22] DEBUG[6868]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  197. [May 21 09:33:22] DEBUG[6868]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  198. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:5775 sip_new: *** Our native formats are 0x8 (alaw)
  199. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:5776 sip_new: *** Joint capabilities are 0x8 (alaw)
  200. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:5777 sip_new: *** Our capabilities are 0x100f (g723|gsm|ulaw|alaw|g722)
  201. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:5778 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
  202. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:5808 sip_new: This channel will not be able to handle video.
  203. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:10853 build_route: build_route: Contact hop: <sip:[email protected]:5061>
  204. list_route: hop: <sip:[email protected]:5061>
  205. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:18349 handle_request_invite: SIP/10000-08226420: New call is still down.... Trying...
  206.  
  207. <--- Transmitting (NAT) to 78.105.1.131:5061 --->
  208. SIP/2.0 100 Trying
  209. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKyinmzjzo;received=78.105.1.131;rport=5061
  210. From: "10000" <sip:[email protected]>;tag=njogt
  211. To: <sip:[email protected]>
  212. Call-ID: [email protected]
  213. CSeq: 866 INVITE
  214. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  215. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  216. Supported: replaces, timer
  217. Contact: <sip:[email protected]>
  218. Content-Length: 0
  219.  
  220.  
  221. <------------>
  222. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 10' onto UDP socket destined for 78.105.1.131:5061
  223. [May 21 09:33:22] DEBUG[6865]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  224. [May 21 09:33:22] DEBUG[6865]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  225. [May 21 09:33:22] DEBUG[6865]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  226. [May 21 09:33:22] DEBUG[6865]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  227. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  228. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  229. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  230. [May 21 09:33:22] DEBUG[6872]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  231. [May 21 09:33:22] DEBUG[6872]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  232. [May 21 09:33:22] DEBUG[6872]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  233. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  234. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  235. [May 21 09:33:22] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  236. [May 21 09:33:22] DEBUG[6872]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- 501
  237. [May 21 09:33:22] DEBUG[6872]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  238. [May 21 09:33:22] DEBUG[6872]: pbx.c:1359 new_find_extension: returning an exact match-- 501
  239. [May 21 09:33:22] DEBUG[6872]: pbx.c:3179 pbx_extension_helper: Launching 'MeetMe'
  240. -- Executing [501@common:1] MeetMe("SIP/10000-08226420", "12,MI") in new stack
  241. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:5347 sip_answer: SIP answering channel: SIP/10000-08226420
  242. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:9013 transmit_response_with_sdp: Setting framing from config on incoming call
  243. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:8681 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True
  244. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:8682 add_sdp: ** Our prefcodec: 0x0 (nothing)
  245. Audio is at 78.105.1.127 port 11092
  246. Adding codec 0x8 (alaw) to SDP
  247. Adding non-codec 0x1 (telephone-event) to SDP
  248. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:8830 add_sdp: -- Done with adding codecs to SDP
  249. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:8949 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw)
  250.  
  251. <--- Reliably Transmitting (NAT) to 78.105.1.131:5061 --->
  252. SIP/2.0 200 OK
  253. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKyinmzjzo;received=78.105.1.131;rport=5061
  254. From: "10000" <sip:[email protected]>;tag=njogt
  255. To: <sip:[email protected]>;tag=as77f937f0
  256. Call-ID: [email protected]
  257. CSeq: 866 INVITE
  258. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  259. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  260. Supported: replaces, timer
  261. Contact: <sip:[email protected]>
  262. Content-Type: application/sdp
  263. Content-Length: 281
  264.  
  265. v=0
  266. o=root 1629583709 1629583709 IN IP4 78.105.1.127
  267. s=Asterisk PBX SVN-branch-1.6.1-r190371M
  268. c=IN IP4 78.105.1.127
  269. t=0 0
  270. m=audio 11092 RTP/AVP 8 101
  271. a=rtpmap:8 PCMA/8000
  272. a=rtpmap:101 telephone-event/8000
  273. a=fmtp:101 0-16
  274. a=silenceSupp:off - - - -
  275. a=ptime:20
  276. a=sendrecv
  277.  
  278. <------------>
  279. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:3200 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #9
  280. [May 21 09:33:22] DEBUG[6872]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 78.105.1.131:5061
  281. [May 21 09:33:22] DEBUG[6865]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  282. [May 21 09:33:22] DEBUG[6865]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  283. [May 21 09:33:22] DEBUG[6865]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  284. [May 21 09:33:22] DEBUG[6865]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  285.  
  286. <--- SIP read from UDP://78.105.1.131:5061 --->
  287. ACK sip:[email protected] SIP/2.0
  288. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKuotdpzum
  289. Max-Forwards: 70
  290. To: <sip:[email protected]>;tag=as77f937f0
  291. From: "10000" <sip:[email protected]>;tag=njogt
  292. Call-ID: [email protected]
  293. CSeq: 866 ACK
  294. Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="7f72e534",uri="sip:[email protected]",response="eae8c2ab46c4ef7f9dc1d97bf2f58ef7",algorithm=MD5
  295. User-Agent: Twinkle/1.1
  296. Content-Length: 0
  297.  
  298.  
  299. <------------->
  300. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 0 [ 32]: ACK sip:[email protected] SIP/2.0
  301. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKuotdpzum
  302. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  303. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 3 [ 47]: To: <sip:[email protected]>;tag=as77f937f0
  304. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=njogt
  305. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  306. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 6 [ 13]: CSeq: 866 ACK
  307. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 7 [173]: Authorization: Digest username="10000",realm="dev-sip.wima.co.uk",nonce="7f72e534",uri="sip:[email protected]",response="eae8c2ab46c4ef7f9dc1d97bf2f58ef7",algorithm=MD5
  308. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 8 [ 23]: User-Agent: Twinkle/1.1
  309. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 9 [ 17]: Content-Length: 0
  310. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 10 [ 0]:
  311. --- (10 headers 0 lines) ---
  312. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:19645 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  313. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:3339 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9
  314. [May 21 09:33:22] DEBUG[6868]: chan_sip.c:3371 __sip_ack: Stopping retransmission on '[email protected]' of Response 866: Match Found
  315. [May 21 09:33:23] DEBUG[6872]: channel.c:1764 __ast_answer: Didn't receive a media frame from SIP/10000-08226420 within 500 ms of answering. Continuing anyway
  316. [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3030 find_conf: The requested confno is '12'?
  317. == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf
  318. == Found
  319. [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference
  320. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay.
  321. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12'
  322. [May 21 09:33:23] DEBUG[6872]: chan_dahdi.c:9549 dahdi_request: Using channel -2
  323. @@[May 21 09:33:23] DEBUG[6872]: devicestate.c:467 devstate_event: device 'DAHDI/pseudo-1074671116' state '2'
  324. @@[May 21 09:33:23] DEBUG[6872]: channel.c:3475 set_format: Set channel DAHDI/pseudo-1074671116 to read format slin
  325. @@[May 21 09:33:23] DEBUG[6872]: channel.c:3475 set_format: Set channel DAHDI/pseudo-1074671116 to write format slin
  326. @@ -- Created MeetMe conference 1023 for conference '12'
  327. @@[May 21 09:33:23] DEBUG[6872]: app.c:645 __ast_play_and_record: play_and_record: vm-rec-name, /var/spool/asterisk/meetme/meetme-username-12-1, 'sln'
  328. @@[May 21 09:33:23] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format ulaw
  329. @@[May 21 09:33:23] DEBUG[6872]: rtp.c:3723 ast_rtp_write: Ooh, format changed from unknown to alaw
  330. @@[May 21 09:33:23] DEBUG[6872]: rtp.c:3739 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160
  331. @@ -- <SIP/10000-08226420> Playing 'vm-rec-name.ulaw' (language 'en')
  332. @@[May 21 09:33:27] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format alaw
  333. @@[May 21 09:33:27] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format ulaw
  334. @@ -- <SIP/10000-08226420> Playing 'beep.ulaw' (language 'en')
  335. [May 21 09:33:28] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format alaw
  336. [May 21 09:33:28] DEBUG[6872]: app.c:666 __ast_play_and_record: Recording Formats: sfmts=sln
  337. -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-12-1 format: sln, 0x8222468
  338. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  339. [May 21 09:33:29] DEBUG[6872]: rtp.c:877 send_dtmf: Sending dtmf: 35 (#), at 78.105.1.131
  340. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  341. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  342. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  343. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  344. [May 21 09:33:29] DEBUG[6872]: rtp.c:877 send_dtmf: Sending dtmf: 35 (#), at 78.105.1.131
  345. -- User ended message by pressing #
  346. [May 21 09:33:29] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format ulaw
  347. [May 21 09:33:29] DEBUG[6872]: rtp.c:3531 ast_rtp_raw_write: Difference is 12968, ms is 1641
  348. -- <SIP/10000-08226420> Playing 'auth-thankyou.ulaw' (language 'en')
  349. [May 21 09:33:29] DEBUG[6872]: rtp.c:1029 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4)
  350. [May 21 09:33:30] DEBUG[6872]: rtp.c:1187 ast_rtcp_read: Got RTCP report of 76 bytes
  351. [May 21 09:33:31] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format alaw
  352. [May 21 09:33:31] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay.
  353. [May 21 09:33:31] DEBUG[6872]: res_config_mysql.c:630 update_mysql: MySQL RealTime: Update SQL: UPDATE conference SET members = '1' WHERE confno = '12'
  354. [May 21 09:33:31] DEBUG[6872]: res_config_mysql.c:644 update_mysql: MySQL RealTime: Updated 1 rows on table: conference
  355. [May 21 09:33:31] DEBUG[6872]: devicestate.c:467 devstate_event: device 'meetme:12' state '2'
  356. [May 21 09:33:31] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format ulaw
  357. -- <SIP/10000-08226420> Playing 'conf-onlyperson.ulaw' (language 'en')
  358. [May 21 09:33:34] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format alaw
  359. [May 21 09:33:34] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to write format slin
  360. [May 21 09:33:34] DEBUG[6872]: channel.c:3475 set_format: Set channel SIP/10000-08226420 to read format slin
  361. [May 21 09:33:34] DEBUG[6872]: app_meetme.c:2090 conf_run: Placed channel SIP/10000-08226420 in DAHDI conf 1023
  362. -- Started music on hold, class 'default', on SIP/10000-08226420
  363. [May 21 09:33:34] DEBUG[6872]: res_musiconhold.c:266 ast_moh_files_next: SIP/10000-08226420 Opened file 1 '/var/lib/asterisk/moh/fpm-sunshine'
  364. [May 21 09:33:34] DEBUG[6872]: rtp.c:3531 ast_rtp_raw_write: Difference is 1120, ms is 160
  365.  
  366. <--- SIP read from UDP://78.105.1.131:5061 --->
  367. BYE sip:[email protected] SIP/2.0
  368. Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKbllmxkti
  369. Max-Forwards: 70
  370. To: <sip:[email protected]>;tag=as77f937f0
  371. From: "10000" <sip:[email protected]>;tag=njogt
  372. Call-ID: [email protected]
  373. CSeq: 867 BYE
  374. User-Agent: Twinkle/1.1
  375. Content-Length: 0
  376.  
  377.  
  378. <------------->
  379. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 0 [ 32]: BYE sip:[email protected] SIP/2.0
  380. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKbllmxkti
  381. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 2 [ 16]: Max-Forwards: 70
  382. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 3 [ 47]: To: <sip:[email protected]>;tag=as77f937f0
  383. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 4 [ 54]: From: "10000" <sip:[email protected]>;tag=njogt
  384. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 5 [ 37]: Call-ID: [email protected]
  385. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 6 [ 13]: CSeq: 867 BYE
  386. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 7 [ 23]: User-Agent: Twinkle/1.1
  387. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 8 [ 17]: Content-Length: 0
  388. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:6691 parse_request: Header 9 [ 0]:
  389. --- (9 headers 0 lines) ---
  390. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:19645 handle_incoming: **** Received BYE (8) - Command in SIP BYE
  391. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:19063 handle_request_bye: Initializing initreq for method BYE - callid [email protected]
  392. Sending to 78.105.1.131 : 5061 (NAT)
  393. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:2645 sip_alreadygone: Setting SIP_ALREADYGONE on dialog [email protected]
  394. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:19146 handle_request_bye: Received bye, issuing owner hangup
  395.  
  396. <--- Transmitting (NAT) to 78.105.1.131:5061 --->
  397. SIP/2.0 200 OK
  398. Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKbllmxkti;received=78.105.1.131;rport=5061
  399. From: "10000" <sip:[email protected]>;tag=njogt
  400. To: <sip:[email protected]>;tag=as77f937f0
  401. Call-ID: [email protected]
  402. CSeq: 867 BYE
  403. Server: Asterisk PBX SVN-branch-1.6.1-r190371M
  404. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  405. Supported: replaces, timer
  406. Content-Length: 0
  407.  
  408.  
  409. <------------>
  410. [May 21 09:33:36] DEBUG[6868]: chan_sip.c:2870 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 78.105.1.131:5061
  411. -- Stopped music on hold on SIP/10000-08226420
  412. [May 21 09:33:36] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay.
  413. [May 21 09:33:36] DEBUG[6872]: res_config_mysql.c:630 update_mysql: MySQL RealTime: Update SQL: UPDATE conference SET members = '0' WHERE confno = '12'
  414. [May 21 09:33:36] DEBUG[6872]: res_config_mysql.c:644 update_mysql: MySQL RealTime: Updated 1 rows on table: conference
  415. [May 21 09:33:36] DEBUG[6872]: devicestate.c:467 devstate_event: device 'meetme:12' state '1'
  416. [May 21 09:33:36] DEBUG[6872]: channel.c:1549 ast_softhangup_nolock: Soft-Hanging up channel 'DAHDI/pseudo-1074671116'
  417. [May 21 09:33:36] DEBUG[6872]: channel.c:1644 ast_hangup: Hanging up channel 'DAHDI/pseudo-1074671116'
  418. [May 21 09:33:36] DEBUG[6872]: chan_dahdi.c:3381 dahdi_hangup: dahdi_hangup(DAHDI/pseudo-1074671116)
  419. [May 21 09:33:36] DEBUG[6872]: chan_dahdi.c:3415 dahdi_hangup: Hangup: channel: -2 index = 0, normal = 24, callwait = -1, thirdcall = -1
  420. [May 21 09:33:36] DEBUG[6872]: chan_dahdi.c:3888 dahdi_setoption: Set option TDD MODE, value: OFF(0) on DAHDI/pseudo-1074671116
  421. [May 21 09:33:36] DEBUG[6872]: chan_dahdi.c:2000 update_conf: Updated conferencing on -2, with 0 conference users
  422. -- Hungup 'DAHDI/pseudo-1074671116'
  423. [May 21 09:33:36] DEBUG[6865]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for DAHDI - pseudo
  424. [May 21 09:33:36] DEBUG[6865]: devicestate.c:487 do_state_change: Changing state for DAHDI/pseudo - state 0 (Unknown)
  425. [May 21 09:33:36] DEBUG[6865]: devicestate.c:467 devstate_event: device 'DAHDI/pseudo' state '0'
  426. [May 21 09:33:36] DEBUG[6872]: pbx.c:3779 __ast_pbx_run: Spawn extension (common,501,1) exited non-zero on 'SIP/10000-08226420'
  427. == Spawn extension (common, 501, 1) exited non-zero on 'SIP/10000-08226420'
  428. [May 21 09:33:36] DEBUG[6872]: channel.c:1549 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/10000-08226420'
  429. [May 21 09:33:36] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  430. [May 21 09:33:36] DEBUG[6872]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- h
  431. [May 21 09:33:36] DEBUG[6872]: pbx.c:1357 new_find_extension: Nothing strange about this match
  432. [May 21 09:33:36] DEBUG[6872]: pbx.c:1358 new_find_extension: returning an exact match-- first found-- h
  433. [May 21 09:33:36] DEBUG[6872]: pbx.c:3179 pbx_extension_helper: Launching 'Hangup'
  434. -- Executing [h@common:1] Hangup("SIP/10000-08226420", "") in new stack
  435. [May 21 09:33:36] DEBUG[6872]: pbx.c:3922 __ast_pbx_run: Spawn extension (common,h,1) exited non-zero on 'SIP/10000-08226420'
  436. == Spawn extension (common, h, 1) exited non-zero on 'SIP/10000-08226420'
  437. [May 21 09:33:36] DEBUG[6872]: channel.c:1644 ast_hangup: Hanging up channel 'SIP/10000-08226420'
  438. [May 21 09:33:36] DEBUG[6872]: chan_sip.c:5176 sip_hangup: Hangup call SIP/10000-08226420, SIP callid [email protected]
  439. [May 21 09:33:36] DEBUG[6872]: cdr.c:1209 ast_cdr_detach: Dropping CDR !
  440. [May 21 09:33:36] DEBUG[6865]: devicestate.c:366 _ast_device_state: No provider found, checking channel drivers for SIP - 10000
  441. [May 21 09:33:36] DEBUG[6865]: chan_sip.c:20925 sip_devicestate: Checking device state for peer 10000
  442. [May 21 09:33:36] DEBUG[6865]: devicestate.c:487 do_state_change: Changing state for SIP/10000 - state 1 (Not in use)
  443. [May 21 09:33:36] DEBUG[6865]: devicestate.c:467 devstate_event: device 'SIP/10000' state '1'
  444. [May 21 09:33:37] DEBUG[6868]: chan_sip.c:4962 sip_destroy: Destroying SIP dialog [email protected]
  445. Really destroying SIP dialog '[email protected]' Method: BYE
  446.  
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