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  1. localhost*CLI>
  2. localhost*CLI>
  3. <--- SIP read from UDP://86.53.0.135:5060 --->
  4.  
  5. <------------->
  6. localhost*CLI>
  7. <--- SIP read from UDP://64.154.41.150:5060 --->
  8. INVITE sip:5854928868@68.44.81.246 SIP/2.0
  9. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;rport
  10. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  11. To: <sip:5854928868@68.44.81.246>
  12. Contact: <sip:6465735875@64.154.41.150>
  13. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  14. CSeq: 102 INVITE
  15. User-Agent: Asterisk PBX
  16. Max-Forwards: 70
  17. Date: Thu, 08 Apr 2010 02:14:51 GMT
  18. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  19. Supported: replaces
  20. Content-Type: application/sdp
  21. Content-Length: 311
  22.  
  23. v=0
  24. o=root 3339 3339 IN IP4 64.154.41.150
  25. s=session
  26. c=IN IP4 64.154.41.150
  27. t=0 0
  28. m=audio 15118 RTP/AVP 0 8 18 100
  29. a=rtpmap:0 PCMU/8000
  30. a=rtpmap:8 PCMA/8000
  31. a=rtpmap:18 G729/8000
  32. a=fmtp:18 annexb=no
  33. a=rtpmap:100 telephone-event/8000
  34. a=fmtp:100 0-16
  35. a=silenceSupp:off - - - -
  36. a=ptime:20
  37. a=sendrecv
  38.  
  39. <------------->
  40. --- (14 headers 15 lines) ---
  41. == Using SIP RTP CoS mark 5
  42. Sending to 64.154.41.150 : 5060 (no NAT)
  43. Using INVITE request as basis request - 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  44. No user '6465735875' in SIP users list
  45. Found peer 'ipcom_incoming' for '6465735875' from 64.154.41.150:5060
  46. Found RTP audio format 0
  47. Found RTP audio format 8
  48. Found RTP audio format 18
  49. Found RTP audio format 100
  50. Found audio description format PCMU for ID 0
  51. Found audio description format PCMA for ID 8
  52. Found audio description format G729 for ID 18
  53. Found audio description format telephone-event for ID 100
  54. Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  55. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  56. Peer audio RTP is at port 64.154.41.150:15118
  57. Looking for 5854928868 in incoming_calls (domain 68.44.81.246)
  58. list_route: hop: <sip:6465735875@64.154.41.150>
  59.  
  60. <--- Transmitting (no NAT) to 64.154.41.150:5060 --->
  61. SIP/2.0 100 Trying
  62. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;received=64.154.41.150;rport=5060
  63. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  64. To: <sip:5854928868@68.44.81.246>
  65. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  66. CSeq: 102 INVITE
  67. User-Agent: Asterisk PBX 1.6.0.26
  68. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  69. Supported: replaces, timer
  70. Contact: <sip:5854928868@192.168.1.158>
  71. Content-Length: 0
  72.  
  73.  
  74. <------------>
  75. -- Executing [5854928868@incoming_calls:1] GotoIf("SIP/ipcom_incoming-0000000a", "1?disa") in new stack
  76. -- Goto (incoming_calls,5854928868,3)
  77. -- Executing [5854928868@incoming_calls:3] Answer("SIP/ipcom_incoming-0000000a", "") in new stack
  78. Audio is at 192.168.1.158 port 18332
  79. Adding codec 0x4 (ulaw) to SDP
  80. Adding non-codec 0x1 (telephone-event) to SDP
  81.  
  82. <--- Reliably Transmitting (no NAT) to 64.154.41.150:5060 --->
  83. SIP/2.0 200 OK
  84. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;received=64.154.41.150;rport=5060
  85. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  86. To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
  87. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  88. CSeq: 102 INVITE
  89. User-Agent: Asterisk PBX 1.6.0.26
  90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  91. Supported: replaces, timer
  92. Contact: <sip:5854928868@192.168.1.158>
  93. Content-Type: application/sdp
  94. Content-Length: 266
  95.  
  96. v=0
  97. o=root 1978871929 1978871929 IN IP4 192.168.1.158
  98. s=Asterisk PBX 1.6.0.26
  99. c=IN IP4 192.168.1.158
  100. t=0 0
  101. m=audio 18332 RTP/AVP 0 100
  102. a=rtpmap:0 PCMU/8000
  103. a=rtpmap:100 telephone-event/8000
  104. a=fmtp:100 0-16
  105. a=silenceSupp:off - - - -
  106. a=ptime:20
  107. a=sendrecv
  108.  
  109. <------------>
  110. localhost*CLI>
  111. <--- SIP read from UDP://64.154.41.150:5060 --->
  112. ACK sip:5854928868@192.168.1.158 SIP/2.0
  113. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK17818e22;rport
  114. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  115. To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
  116. Contact: <sip:6465735875@64.154.41.150>
  117. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  118. CSeq: 102 ACK
  119. User-Agent: Asterisk PBX
  120. Max-Forwards: 70
  121. Content-Length: 0
  122.  
  123.  
  124. <------------->
  125. --- (10 headers 0 lines) ---
  126. -- Executing [5854928868@incoming_calls:4] Set("SIP/ipcom_incoming-0000000a", "TIMEOUT(digit)=5") in new stack
  127. -- Digit timeout set to 5
  128. -- Executing [5854928868@incoming_calls:5] Set("SIP/ipcom_incoming-0000000a", "TIMEOUT(response)=9") in new stack
  129. -- Response timeout set to 9
  130. -- Executing [5854928868@incoming_calls:6] Authenticate("SIP/ipcom_incoming-0000000a", "123") in new stack
  131. -- <SIP/ipcom_incoming-0000000a> Playing 'agent-pass.gsm' (language 'en')
  132. -- <SIP/ipcom_incoming-0000000a> Playing 'auth-thankyou.gsm' (language 'en')
  133. -- Executing [5854928868@incoming_calls:7] DISA("SIP/ipcom_incoming-0000000a", "no-password,outcontext") in new stack
  134. -- Executing [01117327896169@outcontext:1] Dial("SIP/ipcom_incoming-0000000a", "SIP/ipc_outgoing/01117327896169") in new stack
  135. == Using SIP RTP CoS mark 5
  136. Audio is at 192.168.1.158 port 15464
  137. Adding codec 0x4 (ulaw) to SDP
  138. Adding non-codec 0x1 (telephone-event) to SDP
  139. Reliably Transmitting (no NAT) to 86.53.0.135:5060:
  140. INVITE sip:01117327896169@sipagate.com SIP/2.0
  141. Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport
  142. Max-Forwards: 70
  143. From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
  144. To: <sip:01117327896169@sipagate.com>
  145. Contact: <sip:56000351@192.168.1.158>
  146. Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
  147. CSeq: 102 INVITE
  148. User-Agent: Asterisk PBX 1.6.0.26
  149. Date: Thu, 08 Apr 2010 02:15:10 GMT
  150. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  151. Supported: replaces, timer
  152. Content-Type: application/sdp
  153. Content-Length: 266
  154.  
  155. v=0
  156. o=root 1266698409 1266698409 IN IP4 192.168.1.158
  157. s=Asterisk PBX 1.6.0.26
  158. c=IN IP4 192.168.1.158
  159. t=0 0
  160. m=audio 15464 RTP/AVP 0 101
  161. a=rtpmap:0 PCMU/8000
  162. a=rtpmap:101 telephone-event/8000
  163. a=fmtp:101 0-16
  164. a=silenceSupp:off - - - -
  165. a=ptime:20
  166. a=sendrecv
  167.  
  168. ---
  169. -- Called ipc_outgoing/01117327896169
  170. localhost*CLI>
  171. <--- SIP read from UDP://86.53.0.135:5060 --->
  172. SIP/2.0 401 Unauthorized
  173. Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport=1025;received=68.44.81.246
  174. From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
  175. To: <sip:01117327896169@sipagate.com>;tag=4938594e7c94d95c79aee0bf2e224375.96f7
  176. Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
  177. CSeq: 102 INVITE
  178. WWW-Authenticate: Digest realm="sipagate.com", nonce="4bbd3a42a324c5cd2241544ee91217a49f6ac865"
  179. Server: Sip EXpress router (0.9.6 (i386/linux))
  180. Content-Length: 0
  181.  
  182.  
  183. <------------->
  184. --- (9 headers 0 lines) ---
  185. Transmitting (no NAT) to 86.53.0.135:5060:
  186. ACK sip:01117327896169@sipagate.com SIP/2.0
  187. Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport
  188. Max-Forwards: 70
  189. From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
  190. To: <sip:01117327896169@sipagate.com>;tag=4938594e7c94d95c79aee0bf2e224375.96f7
  191. Contact: <sip:56000351@192.168.1.158>
  192. Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
  193. CSeq: 102 ACK
  194. User-Agent: Asterisk PBX 1.6.0.26
  195. Content-Length: 0
  196.  
  197.  
  198. ---
  199. [Apr 7 22:15:10] NOTICE[22058]: chan_sip.c:16131 handle_response_invite: Failed to authenticate on INVITE to '"6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc'
  200. -- SIP/ipc_outgoing-0000000b is circuit-busy
  201. == Everyone is busy/congested at this time (1:0/1/0)
  202. -- Auto fallthrough, channel 'SIP/ipcom_incoming-0000000a' status is 'CONGESTION'
  203. Really destroying SIP dialog '3f9b887d306591ca74fdbf8b18c203ca@sipagate.com' Method: INVITE
  204. Really destroying SIP dialog '1ef7f81d0e715cdb063e1a564a9573dd@127.0.0.1' Method: REGISTER
  205. Really destroying SIP dialog '27a2531352c461a258534e476fff0605@127.0.0.1' Method: REGISTER
  206. localhost*CLI>
  207. <--- SIP read from UDP://64.154.41.150:5060 --->
  208. BYE sip:5854928868@192.168.1.158 SIP/2.0
  209. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK091b07b8;rport
  210. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  211. To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
  212. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  213. CSeq: 103 BYE
  214. User-Agent: Asterisk PBX
  215. Max-Forwards: 70
  216. X-Asterisk-HangupCause: Normal Clearing
  217. X-Asterisk-HangupCauseCode: 16
  218. Content-Length: 0
  219.  
  220.  
  221. <------------->
  222. --- (11 headers 0 lines) ---
  223. Sending to 64.154.41.150 : 5060 (no NAT)
  224.  
  225. <--- Transmitting (no NAT) to 64.154.41.150:5060 --->
  226. SIP/2.0 200 OK
  227. Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK091b07b8;received=64.154.41.150;rport=5060
  228. From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
  229. To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
  230. Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
  231. CSeq: 103 BYE
  232. User-Agent: Asterisk PBX 1.6.0.26
  233. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  234. Supported: replaces, timer
  235. Content-Length: 0
  236.  
  237.  
  238. <------------>
  239. localhost*CLI>
  240. <--- SIP read from UDP://86.53.0.135:5060 --->
  241.  
  242. <------------->
  243. Really destroying SIP dialog '457a25bd233a60d9020e31ff3b925e2c@64.154.41.150' Method: BYE
  244. localhost*CLI>
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