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  1. flat*CLI>
  2. Audio is at 5060
  3. Adding codec 0x8 (alaw) to SDP
  4. Adding codec 0x4 (ulaw) to SDP
  5. Adding codec 0x400 (ilbc) to SDP
  6. Adding non-codec 0x1 (telephone-event) to SDP
  7. Reliably Transmitting (NAT) to 194.120.0.198:5060:
  8. INVITE sip:79876543210@sip.nonoh.net SIP/2.0
  9. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK2ee8d57a;rport
  10. Max-Forwards: 70
  11. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  12. To: <sip:79876543210@sip.nonoh.net>
  13. Contact: <sip:nonohuser@188.14.73.15:5060>
  14. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  15. CSeq: 102 INVITE
  16. User-Agent: Asterisk PBX 1.8.2.2
  17. Date: Wed, 26 Jan 2011 23:01:54 GMT
  18. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  19. Supported: replaces, timer
  20. Content-Type: application/sdp
  21. Content-Length: 336
  22.  
  23. v=0
  24. o=root 1941405858 1941405858 IN IP4 188.14.73.15
  25. s=Asterisk PBX 1.8.2.2
  26. c=IN IP4 188.14.73.15
  27. t=0 0
  28. m=audio 10206 RTP/AVP 8 0 97 101
  29. a=rtpmap:8 PCMA/8000
  30. a=rtpmap:0 PCMU/8000
  31. a=rtpmap:97 iLBC/8000
  32. a=fmtp:97 mode=30
  33. a=rtpmap:101 telephone-event/8000
  34. a=fmtp:101 0-16
  35. a=silenceSupp:off - - - -
  36. a=ptime:20
  37. a=sendrecv
  38.  
  39. ---
  40.  
  41. <--- SIP read from UDP:194.120.0.198:5060 --->
  42. SIP/2.0 401 Unauthorized
  43. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK2ee8d57a;rport
  44. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  45. To: <sip:79876543210@sip.nonoh.net>
  46. Contact: sip:79876543210@194.120.0.198:5060
  47. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  48. CSeq: 102 INVITE
  49. Server: (Very nice Sip Registrar/Proxy Server)
  50. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  51. WWW-Authenticate: Digest realm="sipdiscount.com",nonce="1281676641",algorithm=MD5
  52. Content-Length: 0
  53.  
  54. <------------->
  55. --- (11 headers 0 lines) ---
  56. Transmitting (NAT) to 194.120.0.198:5060:
  57. ACK sip:79876543210@sip.nonoh.net SIP/2.0
  58. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK2ee8d57a;rport
  59. Max-Forwards: 70
  60. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  61. To: <sip:79876543210@sip.nonoh.net>
  62. Contact: <sip:nonohuser@188.14.73.15:5060>
  63. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  64. CSeq: 102 ACK
  65. User-Agent: Asterisk PBX 1.8.2.2
  66. Content-Length: 0
  67.  
  68.  
  69. ---
  70. Audio is at 5060
  71. Adding codec 0x8 (alaw) to SDP
  72. Adding codec 0x4 (ulaw) to SDP
  73. Adding codec 0x400 (ilbc) to SDP
  74. Adding non-codec 0x1 (telephone-event) to SDP
  75. Reliably Transmitting (NAT) to 194.120.0.198:5060:
  76. INVITE sip:79876543210@sip.nonoh.net SIP/2.0
  77. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK1a2f8671;rport
  78. Max-Forwards: 70
  79. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  80. To: <sip:79876543210@sip.nonoh.net>
  81. Contact: <sip:nonohuser@188.14.73.15:5060>
  82. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  83. CSeq: 103 INVITE
  84. User-Agent: Asterisk PBX 1.8.2.2
  85. Authorization: Digest username="nonohuser", realm="sipdiscount.com", algorithm=MD5, uri="sip:79876543210@sip.nonoh.net", nonce="1281676641", response="62589fb25f683b88ac658fad7bb7415e"
  86. Date: Wed, 26 Jan 2011 23:01:54 GMT
  87. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  88. Supported: replaces, timer
  89. Content-Type: application/sdp
  90. Content-Length: 336
  91.  
  92. v=0
  93. o=root 1941405858 1941405859 IN IP4 188.14.73.15
  94. s=Asterisk PBX 1.8.2.2
  95. c=IN IP4 188.14.73.15
  96. t=0 0
  97. m=audio 10206 RTP/AVP 8 0 97 101
  98. a=rtpmap:8 PCMA/8000
  99. a=rtpmap:0 PCMU/8000
  100. a=rtpmap:97 iLBC/8000
  101. a=fmtp:97 mode=30
  102. a=rtpmap:101 telephone-event/8000
  103. a=fmtp:101 0-16
  104. a=silenceSupp:off - - - -
  105. a=ptime:20
  106. a=sendrecv
  107.  
  108. ---
  109. Retransmitting #1 (NAT) to 194.120.0.198:5060:
  110. INVITE sip:79876543210@sip.nonoh.net SIP/2.0
  111. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK1a2f8671;rport
  112. Max-Forwards: 70
  113. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  114. To: <sip:79876543210@sip.nonoh.net>
  115. Contact: <sip:nonohuser@188.14.73.15:5060>
  116. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  117. CSeq: 103 INVITE
  118. User-Agent: Asterisk PBX 1.8.2.2
  119. Authorization: Digest username="nonohuser", realm="sipdiscount.com", algorithm=MD5, uri="sip:79876543210@sip.nonoh.net", nonce="1281676641", response="62589fb25f683b88ac658fad7bb7415e"
  120. Date: Wed, 26 Jan 2011 23:01:54 GMT
  121. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  122. Supported: replaces, timer
  123. Content-Type: application/sdp
  124. Content-Length: 336
  125.  
  126. v=0
  127. o=root 1941405858 1941405859 IN IP4 188.14.73.15
  128. s=Asterisk PBX 1.8.2.2
  129. c=IN IP4 188.14.73.15
  130. t=0 0
  131. m=audio 10206 RTP/AVP 8 0 97 101
  132. a=rtpmap:8 PCMA/8000
  133. a=rtpmap:0 PCMU/8000
  134. a=rtpmap:97 iLBC/8000
  135. a=fmtp:97 mode=30
  136. a=rtpmap:101 telephone-event/8000
  137. a=fmtp:101 0-16
  138. a=silenceSupp:off - - - -
  139. a=ptime:20
  140. a=sendrecv
  141.  
  142. ---
  143.  
  144. <--- SIP read from UDP:194.120.0.198:5060 --->
  145. SIP/2.0 100 Trying
  146. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK1a2f8671;rport
  147. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  148. To: <sip:79876543210@sip.nonoh.net>
  149. Contact: sip:79876543210@194.120.0.198:5060
  150. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  151. CSeq: 103 INVITE
  152. Server: (Very nice Sip Registrar/Proxy Server)
  153. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  154. Content-Length: 0
  155.  
  156. <------------->
  157. --- (10 headers 0 lines) ---
  158.  
  159. <--- SIP read from UDP:194.120.0.198:5060 --->
  160. SIP/2.0 183 Session progress
  161. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK1a2f8671;rport
  162. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  163. To: <sip:79876543210@sip.nonoh.net>;tag=c11710acc12b10ac4d2d9fc22825e1
  164. Contact: sip:79876543210@194.120.0.198:5060
  165. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  166. CSeq: 103 INVITE
  167. Server: (Very nice Sip Registrar/Proxy Server)
  168. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  169. Content-Type: application/sdp
  170. Content-Length: 207
  171.  
  172. v=0
  173. o=nonohuser 1296082916 1296082916 IN IP4 77.72.168.13
  174. s=SIP Call
  175. c=IN IP4 77.72.168.13
  176. t=0 0
  177. m=audio 57484 RTP/AVP 8 101
  178. a=rtpmap:8 PCMA/8000
  179. a=rtpmap:101 telephone-event/8000
  180. a=ptime:20
  181. <------------->
  182. --- (11 headers 9 lines) ---
  183. Found RTP audio format 8
  184. Found RTP audio format 101
  185. Found audio description format PCMA for ID 8
  186. Found audio description format telephone-event for ID 101
  187. Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  188. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  189. Peer audio RTP is at port 77.72.168.13:57484
  190. flat*CLI>
  191. flat*CLI>
  192. flat*CLI>
  193.  
  194. <--- SIP read from UDP:194.120.0.198:5060 --->
  195. SIP/2.0 200 Ok
  196. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK1a2f8671;rport
  197. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  198. To: <sip:79876543210@sip.nonoh.net>;tag=c11710acc12b10ac4d2d9fc22825e1
  199. Contact: sip:79876543210@194.120.0.198:5060
  200. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  201. CSeq: 103 INVITE
  202. Server: (Very nice Sip Registrar/Proxy Server)
  203. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  204. Content-Type: application/sdp
  205. Content-Length: 207
  206.  
  207. v=0
  208. o=nonohuser 1296082949 1296082949 IN IP4 77.72.168.13
  209. s=SIP Call
  210. c=IN IP4 77.72.168.13
  211. t=0 0
  212. m=audio 57484 RTP/AVP 8 101
  213. a=rtpmap:8 PCMA/8000
  214. a=rtpmap:101 telephone-event/8000
  215. a=ptime:20
  216. <------------->
  217. --- (11 headers 9 lines) ---
  218. Found RTP audio format 8
  219. Found RTP audio format 101
  220. Found audio description format PCMA for ID 8
  221. Found audio description format telephone-event for ID 101
  222. Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  223. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  224. Peer audio RTP is at port 77.72.168.13:57484
  225. list_route: hop: <sip:79876543210@194.120.0.198:5060>
  226. set_destination: Parsing <sip:79876543210@194.120.0.198:5060> for address/port to send to
  227. set_destination: set destination to 194.120.0.198:5060
  228. Transmitting (NAT) to 194.120.0.198:5060:
  229. ACK sip:79876543210@194.120.0.198:5060 SIP/2.0
  230. Via: SIP/2.0/UDP 188.14.73.15:5060;branch=z9hG4bK56a69fd2;rport
  231. Max-Forwards: 70
  232. From: "datacard1" <sip:nonohuser@188.14.73.15>;tag=as5bb73afb
  233. To: <sip:79876543210@sip.nonoh.net>;tag=c11710acc12b10ac4d2d9fc22825e1
  234. Contact: <sip:nonohuser@188.14.73.15:5060>
  235. Call-ID: 680dcf7b075e9ea7018122736761e91e@188.14.73.15:5060
  236. CSeq: 103 ACK
  237. User-Agent: Asterisk PBX 1.8.2.2
  238. Content-Length: 0
  239.  
  240.  
  241. ---
  242. [Jan 27 02:02:27] WARNING[18307]: channel.c:974 channel_indicate: [Datacard/datacard1-010000000c] Don't know how to indicate condition 22
  243. flat*CLI>
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