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  1. <--- SIP read from UDP:192.168.0.36:5060 --->
  2. SIP/2.0 200 OK
  3. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  4. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  5. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  6. CSeq: 102 INVITE
  7. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK39fa9396
  8. Contact: "102" <sip:102@192.168.0.36:5060>
  9. Server: Linksys/SPA942-6.1.5(a)
  10. Content-Length: 204
  11. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  12. Supported: replaces
  13. Content-Type: application/sdp
  14.  
  15. v=0
  16. o=- 47752 47752 IN IP4 192.168.0.36
  17. s=-
  18. c=IN IP4 192.168.0.36
  19. t=0 0
  20. m=audio 16464 RTP/AVP 8 101
  21. a=rtpmap:8 PCMA/8000
  22. a=rtpmap:101 telephone-event/8000
  23. a=fmtp:101 0-15
  24. a=ptime:30
  25. a=sendrecv
  26.  
  27. <------------->
  28. --- (12 headers 11 lines) ---
  29. Found RTP audio format 8
  30. Found RTP audio format 101
  31. Found audio description format PCMA for ID 8
  32. Found audio description format telephone-event for ID 101
  33. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  34. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  35. Peer audio RTP is at port 192.168.0.36:16464
  36. list_route: hop: <sip:102@192.168.0.36:5060>
  37. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  38. set_destination: set destination to 192.168.0.36, port 5060
  39. Transmitting (no NAT) to 192.168.0.36:5060:
  40. ACK sip:102@192.168.0.36:5060 SIP/2.0
  41. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6273f09f;rport
  42. Max-Forwards: 70
  43. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  44. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  45. Contact: <sip:774296413@192.168.0.1>
  46. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  47. CSeq: 102 ACK
  48. User-Agent: Asterisk PBX 1.6.2.9-2
  49. Content-Length: 0
  50.  
  51.  
  52. ---
  53. -- SIP/102-00000005 answered SIP/break.viphone.cz-00000004
  54. -- Native bridging SIP/break.viphone.cz-00000004 and SIP/102-00000005
  55. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  56. set_destination: set destination to 192.168.0.36, port 5060
  57. Audio is at 192.168.0.1 port 15142
  58. Adding codec 0x8 (alaw) to SDP
  59. Adding non-codec 0x1 (telephone-event) to SDP
  60. Reliably Transmitting (no NAT) to 192.168.0.36:5060:
  61. INVITE sip:102@192.168.0.36:5060 SIP/2.0
  62. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3d85893e;rport
  63. Max-Forwards: 70
  64. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  65. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  66. Contact: <sip:774296413@192.168.0.1>
  67. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  68. CSeq: 103 INVITE
  69. User-Agent: Asterisk PBX 1.6.2.9-2
  70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  71. Supported: replaces, timer
  72. X-asterisk-Info: SIP re-invite (External RTP bridge)
  73. Content-Type: application/sdp
  74. Content-Length: 240
  75.  
  76. v=0
  77. o=root 746266163 746266164 IN IP4 193.138.78.103
  78. s=Asterisk PBX 1.6.2.9-2
  79. c=IN IP4 193.138.78.103
  80. t=0 0
  81. m=audio 14388 RTP/AVP 8 101
  82. a=rtpmap:8 PCMA/8000
  83. a=rtpmap:101 telephone-event/8000
  84. a=fmtp:101 0-16
  85. a=ptime:20
  86. a=sendrecv
  87.  
  88. ---
  89.  
  90. <--- SIP read from UDP:192.168.0.36:5060 --->
  91. SIP/2.0 200 OK
  92. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  93. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  94. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  95. CSeq: 103 INVITE
  96. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3d85893e
  97. Contact: "102" <sip:102@192.168.0.36:5060>
  98. Server: Linksys/SPA942-6.1.5(a)
  99. Content-Length: 204
  100. Content-Type: application/sdp
  101.  
  102. v=0
  103. o=- 47752 47753 IN IP4 192.168.0.36
  104. s=-
  105. c=IN IP4 192.168.0.36
  106. t=0 0
  107. m=audio 16464 RTP/AVP 8 101
  108. a=rtpmap:8 PCMA/8000
  109. a=rtpmap:101 telephone-event/8000
  110. a=fmtp:101 0-15
  111. a=ptime:30
  112. a=sendrecv
  113.  
  114. <------------->
  115. --- (10 headers 11 lines) ---
  116. Found RTP audio format 8
  117. Found RTP audio format 101
  118. Found audio description format PCMA for ID 8
  119. Found audio description format telephone-event for ID 101
  120. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  121. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  122. Peer audio RTP is at port 192.168.0.36:16464
  123. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  124. set_destination: set destination to 192.168.0.36, port 5060
  125. Transmitting (no NAT) to 192.168.0.36:5060:
  126. ACK sip:102@192.168.0.36:5060 SIP/2.0
  127. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK21802dc2;rport
  128. Max-Forwards: 70
  129. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  130. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  131. Contact: <sip:774296413@192.168.0.1>
  132. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  133. CSeq: 103 ACK
  134. User-Agent: Asterisk PBX 1.6.2.9-2
  135. Content-Length: 0
  136.  
  137.  
  138. ---
  139. -- Got SIP response 406 "Not Acceptable" back from 193.138.78.103
  140. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  141. set_destination: set destination to 192.168.0.36, port 5060
  142. Audio is at 192.168.0.1 port 15142
  143. Adding codec 0x8 (alaw) to SDP
  144. Adding non-codec 0x1 (telephone-event) to SDP
  145. Reliably Transmitting (no NAT) to 192.168.0.36:5060:
  146. INVITE sip:102@192.168.0.36:5060 SIP/2.0
  147. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35274eea;rport
  148. Max-Forwards: 70
  149. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  150. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  151. Contact: <sip:774296413@192.168.0.1>
  152. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  153. CSeq: 104 INVITE
  154. User-Agent: Asterisk PBX 1.6.2.9-2
  155. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  156. Supported: replaces, timer
  157. X-asterisk-Info: SIP re-invite (External RTP bridge)
  158. Content-Type: application/sdp
  159. Content-Length: 234
  160.  
  161. v=0
  162. o=root 746266163 746266165 IN IP4 192.168.0.1
  163. s=Asterisk PBX 1.6.2.9-2
  164. c=IN IP4 192.168.0.1
  165. t=0 0
  166. m=audio 15142 RTP/AVP 8 101
  167. a=rtpmap:8 PCMA/8000
  168. a=rtpmap:101 telephone-event/8000
  169. a=fmtp:101 0-16
  170. a=ptime:20
  171. a=sendrecv
  172.  
  173. ---
  174.  
  175. <--- SIP read from UDP:192.168.0.36:5060 --->
  176. SIP/2.0 200 OK
  177. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  178. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  179. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  180. CSeq: 104 INVITE
  181. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35274eea
  182. Contact: "102" <sip:102@192.168.0.36:5060>
  183. Server: Linksys/SPA942-6.1.5(a)
  184. Content-Length: 204
  185. Content-Type: application/sdp
  186.  
  187. v=0
  188. o=- 47752 47754 IN IP4 192.168.0.36
  189. s=-
  190. c=IN IP4 192.168.0.36
  191. t=0 0
  192. m=audio 16464 RTP/AVP 8 101
  193. a=rtpmap:8 PCMA/8000
  194. a=rtpmap:101 telephone-event/8000
  195. a=fmtp:101 0-15
  196. a=ptime:30
  197. a=sendrecv
  198.  
  199. <------------->
  200. --- (10 headers 11 lines) ---
  201. Found RTP audio format 8
  202. Found RTP audio format 101
  203. Found audio description format PCMA for ID 8
  204. Found audio description format telephone-event for ID 101
  205. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  206. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  207. Peer audio RTP is at port 192.168.0.36:16464
  208. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  209. set_destination: set destination to 192.168.0.36, port 5060
  210. Transmitting (no NAT) to 192.168.0.36:5060:
  211. ACK sip:102@192.168.0.36:5060 SIP/2.0
  212. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6a918b73;rport
  213. Max-Forwards: 70
  214. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  215. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  216. Contact: <sip:774296413@192.168.0.1>
  217. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  218. CSeq: 104 ACK
  219. User-Agent: Asterisk PBX 1.6.2.9-2
  220. Content-Length: 0
  221.  
  222.  
  223. ---
  224. Scheduling destruction of SIP dialog '749e5e675d398de65fc3e3303dfa03e2@192.168.0.1' in 6400 ms (Method: INVITE)
  225. set_destination: Parsing <sip:102@192.168.0.36:5060> for address/port to send to
  226. set_destination: set destination to 192.168.0.36, port 5060
  227. Reliably Transmitting (no NAT) to 192.168.0.36:5060:
  228. BYE sip:102@192.168.0.36:5060 SIP/2.0
  229. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0d4feb6b;rport
  230. Max-Forwards: 70
  231. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  232. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  233. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  234. CSeq: 105 BYE
  235. User-Agent: Asterisk PBX 1.6.2.9-2
  236. X-Asterisk-HangupCause: Normal Clearing
  237. X-Asterisk-HangupCauseCode: 16
  238. Content-Length: 0
  239.  
  240.  
  241. ---
  242. == Spawn extension (inbound, s, 1) exited non-zero on 'SIP/break.viphone.cz-00000004'
  243.  
  244. <--- SIP read from UDP:192.168.0.36:5060 --->
  245. SIP/2.0 200 OK
  246. To: <sip:102@192.168.0.36:5060>;tag=17e810926e19c59di0
  247. From: "774296413" <sip:774296413@192.168.0.1>;tag=as4000ad19
  248. Call-ID: 749e5e675d398de65fc3e3303dfa03e2@192.168.0.1
  249. CSeq: 105 BYE
  250. Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0d4feb6b
  251. Server: Linksys/SPA942-6.1.5(a)
  252. Content-Length: 0
  253.  
  254.  
  255. <------------->
  256. --- (8 headers 0 lines) ---
  257. Really destroying SIP dialog '749e5e675d398de65fc3e3303dfa03e2@192.168.0.1' Method: INVITE
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