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Jul 14th, 2014
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  1. <--- SIP read from UDP:192.168.1.138:5060 --->
  2. INVITE sip:*2*220000@192.168.1.248 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160
  4. Max-Forwards: 70
  5. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  6. To: <sip:*2*220000@192.168.1.248>
  7. Contact: <sip:805@192.168.1.138:5060>
  8. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  9. CSeq: 31252 INVITE
  10. Subject: Phone call
  11. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
  12. Supported: replaces, 100rel
  13. Content-Type: application/sdp
  14. Content-Length: 405
  15.  
  16. v=0
  17. o=master-pc 3614303752 0 IN IP4 192.168.1.138
  18. s=sflphone
  19. c=IN IP4 192.168.1.138
  20. t=0 0
  21. m=audio 27250 RTP/AVP 0 3 8 9 110 111 112 101
  22. a=rtpmap:0 PCMU/8000
  23. a=rtpmap:3 GSM/8000
  24. a=rtpmap:8 PCMA/8000
  25. a=rtpmap:9 G722/8000
  26. a=rtpmap:110 speex/8000
  27. a=rtpmap:111 speex/16000
  28. a=rtpmap:112 speex/32000
  29. a=sendrecv
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-15
  32. a=rtcp:27251 IN IP4 192.168.1.138
  33. <------------->
  34. --- (13 headers 17 lines) ---
  35. Sending to 192.168.1.138:5060 (no NAT)
  36. Sending to 192.168.1.138:5060 (no NAT)
  37. Using INVITE request as basis request - e1acda66-823b-46f6-9354-4d8bdf3638b0
  38. Found peer '805' for '805' from 192.168.1.138:5060
  39.  
  40. <--- Reliably Transmitting (no NAT) to 192.168.1.138:5060 --->
  41. SIP/2.0 401 Unauthorized
  42. Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160;received=192.168.1.138;rport=5060
  43. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  44. To: <sip:*2*220000@192.168.1.248>;tag=as7568ea5f
  45. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  46. CSeq: 31252 INVITE
  47. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  48. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  49. Supported: replaces, timer
  50. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54556ff6"
  51. Content-Length: 0
  52.  
  53.  
  54. <------------>
  55. Scheduling destruction of SIP dialog 'e1acda66-823b-46f6-9354-4d8bdf3638b0' in 6400 ms (Method: INVITE)
  56.  
  57. <--- SIP read from UDP:192.168.1.138:5060 --->
  58. ACK sip:*2*220000@192.168.1.248 SIP/2.0
  59. Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160
  60. Max-Forwards: 70
  61. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  62. To: <sip:*2*220000@192.168.1.248>;tag=as7568ea5f
  63. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  64. CSeq: 31252 ACK
  65. Content-Length: 0
  66.  
  67. <------------->
  68. --- (8 headers 0 lines) ---
  69.  
  70. <--- SIP read from UDP:192.168.1.138:5060 --->
  71. INVITE sip:*2*220000@192.168.1.248 SIP/2.0
  72. Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3
  73. Max-Forwards: 70
  74. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  75. To: <sip:*2*220000@192.168.1.248>
  76. Contact: <sip:805@192.168.1.138:5060>
  77. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  78. CSeq: 31253 INVITE
  79. Subject: Phone call
  80. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
  81. Supported: replaces, 100rel
  82. Authorization: Digest username="805", realm="asterisk", nonce="54556ff6", uri="sip:*2*220000@192.168.1.248", response="390c7717e5786714118d5bcc22096a34", algorithm=MD5
  83. Content-Type: application/sdp
  84. Content-Length: 405
  85.  
  86. v=0
  87. o=master-pc 3614303752 0 IN IP4 192.168.1.138
  88. s=sflphone
  89. c=IN IP4 192.168.1.138
  90. t=0 0
  91. m=audio 27250 RTP/AVP 0 3 8 9 110 111 112 101
  92. a=rtpmap:0 PCMU/8000
  93. a=rtpmap:3 GSM/8000
  94. a=rtpmap:8 PCMA/8000
  95. a=rtpmap:9 G722/8000
  96. a=rtpmap:110 speex/8000
  97. a=rtpmap:111 speex/16000
  98. a=rtpmap:112 speex/32000
  99. a=sendrecv
  100. a=rtpmap:101 telephone-event/8000
  101. a=fmtp:101 0-15
  102. a=rtcp:27251 IN IP4 192.168.1.138
  103. <------------->
  104. --- (14 headers 17 lines) ---
  105. Sending to 192.168.1.138:5060 (no NAT)
  106. Using INVITE request as basis request - e1acda66-823b-46f6-9354-4d8bdf3638b0
  107. Found peer '805' for '805' from 192.168.1.138:5060
  108. == Using SIP RTP CoS mark 5
  109. Found RTP audio format 0
  110. Found RTP audio format 3
  111. Found RTP audio format 8
  112. Found RTP audio format 9
  113. Found RTP audio format 110
  114. Found RTP audio format 111
  115. Found RTP audio format 112
  116. Found RTP audio format 101
  117. Found audio description format PCMU for ID 0
  118. Found audio description format GSM for ID 3
  119. Found audio description format PCMA for ID 8
  120. Found audio description format G722 for ID 9
  121. Found audio description format speex for ID 110
  122. Found audio description format speex for ID 111
  123. Found audio description format speex for ID 112
  124. Found audio description format telephone-event for ID 101
  125. Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|g722|speex32)/video=(nothing)/text=(nothing), combined - (alaw)
  126. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  127. Peer audio RTP is at port 192.168.1.138:27250
  128. Looking for *2*220000 in office-phones (domain 192.168.1.248)
  129. list_route: hop: <sip:805@192.168.1.138:5060>
  130.  
  131. <--- Transmitting (no NAT) to 192.168.1.138:5060 --->
  132. SIP/2.0 100 Trying
  133. Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3;received=192.168.1.138;rport=5060
  134. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  135. To: <sip:*2*220000@192.168.1.248>
  136. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  137. CSeq: 31253 INVITE
  138. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  139. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  140. Supported: replaces, timer
  141. Contact: <sip:*2*220000@192.168.1.248:5060>
  142. Content-Length: 0
  143.  
  144.  
  145. <------------>
  146. -- Executing [*2*220000@office-phones:1] NoOp("SIP/805-00000002", "") in new stack
  147. -- Executing [*2*220000@office-phones:2] Dial("SIP/805-00000002", "SIP/teledyne/220000") in new stack
  148. == Using SIP RTP CoS mark 5
  149. Audio is at 12140
  150. Adding codec 100004 (alaw) to SDP
  151. Adding codec 100002 (gsm) to SDP
  152. Adding codec 100003 (ulaw) to SDP
  153. Adding codec 100017 (testlaw) to SDP
  154. Reliably Transmitting (no NAT) to 91.142.144.132:5060:
  155. INVITE sip:220000@91.142.144.132 SIP/2.0
  156. Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK335c0318
  157. Max-Forwards: 70
  158. From: <sip:NAME@tsl.ru>;tag=as226b3fd8
  159. To: <sip:220000@91.142.144.132>
  160. Contact: <sip:NAME@192.168.1.248:5060>
  161. Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
  162. CSeq: 102 INVITE
  163. User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  164. Date: Mon, 14 Jul 2014 05:15:42 GMT
  165. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  166. Supported: replaces, timer
  167. Content-Type: application/sdp
  168. Content-Length: 240
  169.  
  170. v=0
  171. o=root 769084428 769084428 IN IP4 192.168.1.248
  172. s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
  173. c=IN IP4 192.168.1.248
  174. t=0 0
  175. m=audio 12140 RTP/AVP 8 3 0
  176. a=rtpmap:8 PCMA/8000
  177. a=rtpmap:3 GSM/8000
  178. a=rtpmap:0 PCMU/8000
  179. a=ptime:20
  180. a=sendrecv
  181.  
  182. ---
  183. -- Called SIP/teledyne/220000
  184.  
  185. <--- SIP read from UDP:91.142.144.132:5060 --->
  186. SIP/2.0 100 Trying
  187. Via: SIP/2.0/UDP 192.168.1.248:5060;received=195.208.35.31;branch=z9hG4bK335c0318;rport=5060
  188. From: <sip:NAME@tsl.ru>;tag=as226b3fd8
  189. To: <sip:220000@91.142.144.132>
  190. Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
  191. CSeq: 102 INVITE
  192.  
  193. <------------->
  194. --- (6 headers 0 lines) ---
  195.  
  196. <--- SIP read from UDP:91.142.144.132:5060 --->
  197. SIP/2.0 604 Does not exist anywhere
  198. Via: SIP/2.0/UDP 192.168.1.248:5060;received=195.208.35.31;branch=z9hG4bK335c0318;rport=5060
  199. From: <sip:NAME@tsl.ru>;tag=as226b3fd8
  200. To: <sip:220000@91.142.144.132>;tag=1877391753-1405314942990
  201. Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
  202. CSeq: 102 INVITE
  203. Content-Length: 0
  204.  
  205. <------------->
  206. --- (7 headers 0 lines) ---
  207. -- Got SIP response 604 "Does not exist anywhere" back from 91.142.144.132:5060
  208. Transmitting (no NAT) to 91.142.144.132:5060:
  209. ACK sip:220000@91.142.144.132 SIP/2.0
  210. Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK335c0318
  211. Max-Forwards: 70
  212. From: <sip:NAME@tsl.ru>;tag=as226b3fd8
  213. To: <sip:220000@91.142.144.132>;tag=1877391753-1405314942990
  214. Contact: <sip:NAME@192.168.1.248:5060>
  215. Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
  216. CSeq: 102 ACK
  217. User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  218. Content-Length: 0
  219.  
  220.  
  221. ---
  222. == Everyone is busy/congested at this time (1:0/0/1)
  223. -- Auto fallthrough, channel 'SIP/805-00000002' status is 'CHANUNAVAIL'
  224.  
  225. <--- Reliably Transmitting (no NAT) to 192.168.1.138:5060 --->
  226. SIP/2.0 503 Service Unavailable
  227. Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3;received=192.168.1.138;rport=5060
  228. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  229. To: <sip:*2*220000@192.168.1.248>;tag=as3d3d78c5
  230. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  231. CSeq: 31253 INVITE
  232. Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
  233. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  234. Supported: replaces, timer
  235. X-Asterisk-HangupCause: Unallocated (unassigned) number
  236. X-Asterisk-HangupCauseCode: 1
  237. Content-Length: 0
  238.  
  239.  
  240. <------------>
  241.  
  242. <--- SIP read from UDP:192.168.1.138:5060 --->
  243. ACK sip:*2*220000@192.168.1.248 SIP/2.0
  244. Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3
  245. Max-Forwards: 70
  246. From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
  247. To: <sip:*2*220000@192.168.1.248>;tag=as3d3d78c5
  248. Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
  249. CSeq: 31253 ACK
  250. Content-Length: 0
  251.  
  252. <------------->
  253. --- (8 headers 0 lines) ---
  254. Really destroying SIP dialog '6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru' Method: INVITE
  255. Really destroying SIP dialog 'e1acda66-823b-46f6-9354-4d8bdf3638b0' Method: ACK
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