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  1. localhost*CLI> channel originate SIP/call-labs/16048622117 application Playback tt-weasels
  2. == Using SIP RTP CoS mark 5
  3. Audio is at 5060
  4. Adding codec 0x4 (ulaw) to SDP
  5. Adding codec 0x2 (gsm) to SDP
  6. Adding codec 0x8 (alaw) to SDP
  7. Adding codec 0x800000000000 (testlaw) to SDP
  8. Adding non-codec 0x1 (telephone-event) to SDP
  9. Reliably Transmitting (NAT) to 67.207.160.50:5060:
  10. INVITE sip:16048622117@67.207.160.50 SIP/2.0
  11. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK32f921e8;rport
  12. Max-Forwards: 70
  13. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  14. To: <sip:16048622117@67.207.160.50>
  15. Contact: <sip:9261105187@192.168.1.149:5060>
  16. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  17. CSeq: 102 INVITE
  18. User-Agent: Asterisk PBX 1.8.8.1
  19. Date: Mon, 09 Jan 2012 03:47:24 GMT
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  21. Supported: replaces, timer
  22. Content-Type: application/sdp
  23. Content-Length: 285
  24.  
  25. v=0
  26. o=root 2053730724 2053730724 IN IP4 192.168.1.149
  27. s=Asterisk PBX 1.8.8.1
  28. c=IN IP4 192.168.1.149
  29. t=0 0
  30. m=audio 19214 RTP/AVP 0 3 8 101
  31. a=rtpmap:0 PCMU/8000
  32. a=rtpmap:3 GSM/8000
  33. a=rtpmap:8 PCMA/8000
  34. a=rtpmap:101 telephone-event/8000
  35. a=fmtp:101 0-16
  36. a=ptime:20
  37. a=sendrecv
  38.  
  39. ---
  40.  
  41. <--- SIP read from UDP:67.207.160.50:5060 --->
  42. SIP/2.0 401 Unauthorized
  43. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK32f921e8;received=184.65.142.249;rport=5060
  44. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  45. To: <sip:16048622117@67.207.160.50>;tag=as1dfa95dd
  46. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  47. CSeq: 102 INVITE
  48. Server: Asterisk PBX 1.8.8.0
  49. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  50. Supported: replaces, timer
  51. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="125096e9"
  52. Content-Length: 0
  53.  
  54. <------------->
  55. --- (11 headers 0 lines) ---
  56. Transmitting (NAT) to 67.207.160.50:5060:
  57. ACK sip:16048622117@67.207.160.50 SIP/2.0
  58. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK32f921e8;rport
  59. Max-Forwards: 70
  60. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  61. To: <sip:16048622117@67.207.160.50>;tag=as1dfa95dd
  62. Contact: <sip:9261105187@192.168.1.149:5060>
  63. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  64. CSeq: 102 ACK
  65. User-Agent: Asterisk PBX 1.8.8.1
  66. Content-Length: 0
  67.  
  68.  
  69. ---
  70. Audio is at 5060
  71. Adding codec 0x4 (ulaw) to SDP
  72. Adding codec 0x2 (gsm) to SDP
  73. Adding codec 0x8 (alaw) to SDP
  74. Adding codec 0x800000000000 (testlaw) to SDP
  75. Adding non-codec 0x1 (telephone-event) to SDP
  76. Reliably Transmitting (NAT) to 67.207.160.50:5060:
  77. INVITE sip:16048622117@67.207.160.50 SIP/2.0
  78. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK5cc28470;rport
  79. Max-Forwards: 70
  80. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  81. To: <sip:16048622117@67.207.160.50>
  82. Contact: <sip:9261105187@192.168.1.149:5060>
  83. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  84. CSeq: 103 INVITE
  85. User-Agent: Asterisk PBX 1.8.8.1
  86. Authorization: Digest username="9261105187", realm="asterisk", algorithm=MD5, uri="sip:16048622117@67.207.160.50", nonce="125096e9", response="edd51f7b1abc9b9f309ebf9a2a61812d"
  87. Date: Mon, 09 Jan 2012 03:47:24 GMT
  88. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  89. Supported: replaces, timer
  90. Content-Type: application/sdp
  91. Content-Length: 285
  92.  
  93. v=0
  94. o=root 2053730724 2053730725 IN IP4 192.168.1.149
  95. s=Asterisk PBX 1.8.8.1
  96. c=IN IP4 192.168.1.149
  97. t=0 0
  98. m=audio 19214 RTP/AVP 0 3 8 101
  99. a=rtpmap:0 PCMU/8000
  100. a=rtpmap:3 GSM/8000
  101. a=rtpmap:8 PCMA/8000
  102. a=rtpmap:101 telephone-event/8000
  103. a=fmtp:101 0-16
  104. a=ptime:20
  105. a=sendrecv
  106.  
  107. ---
  108.  
  109. <--- SIP read from UDP:67.207.160.50:5060 --->
  110. SIP/2.0 100 Trying
  111. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK5cc28470;received=184.65.142.249;rport=5060
  112. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  113. To: <sip:16048622117@67.207.160.50>
  114. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  115. CSeq: 103 INVITE
  116. Server: Asterisk PBX 1.8.8.0
  117. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  118. Supported: replaces, timer
  119. Contact: <sip:16048622117@67.207.160.50:5060>
  120. Content-Length: 0
  121.  
  122. <------------->
  123. --- (11 headers 0 lines) ---
  124.  
  125. <--- SIP read from UDP:67.207.160.50:5060 --->
  126. SIP/2.0 200 OK
  127. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK5cc28470;received=184.65.142.249;rport=5060
  128. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  129. To: <sip:16048622117@67.207.160.50>;tag=as0f283ab7
  130. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  131. CSeq: 103 INVITE
  132. Server: Asterisk PBX 1.8.8.0
  133. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  134. Supported: replaces, timer
  135. Contact: <sip:16048622117@67.207.160.50:5060>
  136. Content-Type: application/sdp
  137. Content-Length: 310
  138.  
  139. v=0
  140. o=root 842192541 842192541 IN IP4 67.207.160.50
  141. s=Asterisk PBX 1.8.8.0
  142. c=IN IP4 67.207.160.50
  143. t=0 0
  144. m=audio 15786 RTP/AVP 0 8 3 101
  145. a=rtpmap:0 PCMU/8000
  146. a=rtpmap:8 PCMA/8000
  147. a=rtpmap:3 GSM/8000
  148. a=rtpmap:101 telephone-event/8000
  149. a=fmtp:101 0-16
  150. a=silenceSupp:off - - - -
  151. a=ptime:20
  152. a=sendrecv
  153. <------------->
  154. --- (12 headers 14 lines) ---
  155. Found RTP audio format 0
  156. Found RTP audio format 8
  157. Found RTP audio format 3
  158. Found RTP audio format 101
  159. Found audio description format PCMU for ID 0
  160. Found audio description format PCMA for ID 8
  161. Found audio description format GSM for ID 3
  162. Found audio description format telephone-event for ID 101
  163. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  164. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  165. Peer audio RTP is at port 67.207.160.50:15786
  166. list_route: hop: <sip:16048622117@67.207.160.50:5060>
  167. set_destination: Parsing <sip:16048622117@67.207.160.50:5060> for address/port to send to
  168. set_destination: set destination to 67.207.160.50:5060
  169. Transmitting (NAT) to 67.207.160.50:5060:
  170. ACK sip:16048622117@67.207.160.50:5060 SIP/2.0
  171. Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK43082b1a;rport
  172. Max-Forwards: 70
  173. From: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  174. To: <sip:16048622117@67.207.160.50>;tag=as0f283ab7
  175. Contact: <sip:9261105187@192.168.1.149:5060>
  176. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  177. CSeq: 103 ACK
  178. User-Agent: Asterisk PBX 1.8.8.1
  179. Content-Length: 0
  180.  
  181.  
  182. ---
  183. -- Launching Playback(tt-weasels) on SIP/call-labs-00000057
  184. -- <SIP/call-labs-00000057> Playing 'tt-weasels.ulaw' (language 'en')
  185.  
  186. <--- SIP read from UDP:67.207.160.50:5060 --->
  187. BYE sip:9261105187@192.168.1.149:5060 SIP/2.0
  188. Via: SIP/2.0/UDP 67.207.160.50:5060;branch=z9hG4bK325289e8;rport
  189. Max-Forwards: 70
  190. From: <sip:16048622117@67.207.160.50>;tag=as0f283ab7
  191. To: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  192. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  193. CSeq: 102 BYE
  194. User-Agent: Asterisk PBX 1.8.8.0
  195. Proxy-Authorization: Digest username="9261105187", realm="asterisk", algorithm=MD5, uri="sip:67.207.160.50", nonce="", response="74ca33ffa1b56724a87ba2e9db7434f6"
  196. X-Asterisk-HangupCause: Unknown
  197. X-Asterisk-HangupCauseCode: 0
  198. Content-Length: 0
  199.  
  200. <------------->
  201. --- (12 headers 0 lines) ---
  202. Sending to 67.207.160.50:5060 (NAT)
  203. Scheduling destruction of SIP dialog '11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060' in 32000 ms (Method: BYE)
  204.  
  205. <--- Transmitting (NAT) to 67.207.160.50:5060 --->
  206. SIP/2.0 200 OK
  207. Via: SIP/2.0/UDP 67.207.160.50:5060;branch=z9hG4bK325289e8;received=67.207.160.50;rport=5060
  208. From: <sip:16048622117@67.207.160.50>;tag=as0f283ab7
  209. To: "asterisk" <sip:9261105187@192.168.1.149>;tag=as4cc58c1c
  210. Call-ID: 11d942911411ae5d2f5e46c51ec71126@192.168.1.149:5060
  211. CSeq: 102 BYE
  212. Server: Asterisk PBX 1.8.8.1
  213. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  214. Supported: replaces, timer
  215. Content-Length: 0
  216.  
  217.  
  218. <------------>
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