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  1.  
  2.  
  3. <--- SIP read from UDP:192.168.1.102:5070 --->
  4. INVITE sip:4183172685@192.168.1.207 SIP/2.0
  5. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1856108
  6. To: <sip:4183172685@192.168.1.207>
  7. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  8. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  9. CSeq: 157 INVITE
  10. Max-Forwards: 70
  11. User-Agent: NCH Software Express Talk 4.35
  12. Contact: <sip:0000FFFF0001@192.168.1.102:5070>
  13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  14. Supported: replaces
  15. Content-Type: application/sdp
  16. Content-Length: 354
  17.  
  18. v=0
  19. o=NCHSoftware-Talk 1447514436 1447514459 IN IP4 192.168.1.102
  20. s=Express Talk Call
  21. c=IN IP4 192.168.1.102
  22. t=0 0
  23. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  24. a=rtpmap:0 PCMU/8000
  25. a=rtpmap:8 PCMA/8000
  26. a=rtpmap:96 G726-32/8000
  27. a=rtpmap:3 GSM/8000
  28. a=rtpmap:13 CN/8000
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-16
  31. a=sendrecv
  32. a=direction:active
  33. <------------->
  34. --- (13 headers 15 lines) ---
  35. Sending to 192.168.1.102:5070 (NAT)
  36. Sending to 192.168.1.102:5070 (NAT)
  37. Using INVITE request as basis request - 1447514424-6108-GAMING-PC@192.168.1.102
  38. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  39.  
  40. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  41. SIP/2.0 401 Unauthorized
  42. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1856108;received=192.168.1.102;rport=5070
  43. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  44. To: <sip:4183172685@192.168.1.207>;tag=as3bd8ffd4
  45. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  46. CSeq: 157 INVITE
  47. Server: Asterisk PBX 13.6.0
  48. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  49. Supported: replaces, timer
  50. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bc25834"
  51. Content-Length: 0
  52.  
  53.  
  54. <------------>
  55. Scheduling destruction of SIP dialog '1447514424-6108-GAMING-PC@192.168.1.102' in 32000 ms (Method: INVITE)
  56.  
  57. <--- SIP read from UDP:192.168.1.102:5070 --->
  58. ACK sip:4183172685@192.168.1.207 SIP/2.0
  59. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1856108
  60. To: <sip:4183172685@192.168.1.207>;tag=as3bd8ffd4
  61. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  62. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  63. CSeq: 157 ACK
  64. Max-Forwards: 20
  65. User-Agent: NCH Software Express Talk 4.35
  66. Content-Length: 0
  67.  
  68. <------------->
  69. --- (9 headers 0 lines) ---
  70.  
  71. <--- SIP read from UDP:192.168.1.102:5070 --->
  72. INVITE sip:4183172685@192.168.1.207 SIP/2.0
  73. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1866108
  74. To: <sip:4183172685@192.168.1.207>
  75. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  76. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  77. CSeq: 158 INVITE
  78. Max-Forwards: 20
  79. User-Agent: NCH Software Express Talk 4.35
  80. Contact: <sip:0000FFFF0001@192.168.1.102:5070>
  81. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="5bc25834",uri="sip:4183172685@192.168.1.207",response="d7fdabc74ad635888637b578b9138b27",opaque="",algorithm=MD5
  82. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  83. Supported: replaces
  84. Content-Type: application/sdp
  85. Content-Length: 354
  86.  
  87. v=0
  88. o=NCHSoftware-Talk 1447514436 1447514459 IN IP4 192.168.1.102
  89. s=Express Talk Call
  90. c=IN IP4 192.168.1.102
  91. t=0 0
  92. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  93. a=rtpmap:0 PCMU/8000
  94. a=rtpmap:8 PCMA/8000
  95. a=rtpmap:96 G726-32/8000
  96. a=rtpmap:3 GSM/8000
  97. a=rtpmap:13 CN/8000
  98. a=rtpmap:101 telephone-event/8000
  99. a=fmtp:101 0-16
  100. a=sendrecv
  101. a=direction:active
  102. <------------->
  103. --- (14 headers 15 lines) ---
  104. Sending to 192.168.1.102:5070 (NAT)
  105. Using INVITE request as basis request - 1447514424-6108-GAMING-PC@192.168.1.102
  106. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  107. == Using SIP RTP CoS mark 5
  108. Found RTP audio format 0
  109. Found RTP audio format 8
  110. Found RTP audio format 96
  111. Found RTP audio format 3
  112. Found RTP audio format 13
  113. Found RTP audio format 101
  114. Found audio description format PCMU for ID 0
  115. Found audio description format PCMA for ID 8
  116. Found audio description format G726-32 for ID 96
  117. Found audio description format GSM for ID 3
  118. Found audio description format CN for ID 13
  119. Found audio description format telephone-event for ID 101
  120. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  121. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  122. Peer audio RTP is at port 192.168.1.102:8000
  123. Looking for 4183172685 in LocalSets (domain 192.168.1.207)
  124. sip_route_dump: route/path hop: <sip:0000FFFF0001@192.168.1.102:5070>
  125.  
  126. <--- Transmitting (NAT) to 192.168.1.102:5070 --->
  127. SIP/2.0 100 Trying
  128. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1866108;received=192.168.1.102;rport=5070
  129. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  130. To: <sip:4183172685@192.168.1.207>
  131. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  132. CSeq: 158 INVITE
  133. Server: Asterisk PBX 13.6.0
  134. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  135. Supported: replaces, timer
  136. Contact: <sip:4183172685@192.168.1.207:5060>
  137. Content-Length: 0
  138.  
  139.  
  140. <------------>
  141. -- Executing [4183172685@LocalSets:1] Log("SIP/0000FFFF0001-00000000", "NOTICE, Dialing out from "Stephen" <0000FFFF0001> to 183172685 through Foo Provider") in new stack
  142. -- Executing [4183172685@LocalSets:2] Dial("SIP/0000FFFF0001-00000000", "SIP/4183172685@callcentric") in new stack
  143. == Using SIP RTP CoS mark 5
  144. Audio is at 27464
  145. Adding codec ulaw to SDP
  146. Adding non-codec 0x1 (telephone-event) to SDP
  147. Reliably Transmitting (no NAT) to 204.11.192.39:5080:
  148. INVITE sip:4183172685@callcentric.com SIP/2.0
  149. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK48acd271
  150. Max-Forwards: 70
  151. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  152. To: <sip:4183172685@callcentric.com>
  153. Contact: <sip:17772409788@142.134.91.178:5060>
  154. Call-ID: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  155. CSeq: 102 INVITE
  156. User-Agent: Asterisk PBX 13.6.0
  157. Date: Sat, 14 Nov 2015 20:52:01 GMT
  158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  159. Supported: replaces, timer
  160. Content-Type: application/sdp
  161. Content-Length: 243
  162.  
  163. v=0
  164. o=root 1847464334 1847464334 IN IP4 142.134.91.178
  165. s=Asterisk PBX 13.6.0
  166. c=IN IP4 142.134.91.178
  167. t=0 0
  168. m=audio 27464 RTP/AVP 0 101
  169. a=rtpmap:0 PCMU/8000
  170. a=rtpmap:101 telephone-event/8000
  171. a=fmtp:101 0-16
  172. a=maxptime:150
  173. a=sendrecv
  174.  
  175. ---
  176. -- Called SIP/4183172685@callcentric
  177.  
  178. <--- SIP read from UDP:204.11.192.39:5080 --->
  179. SIP/2.0 407 Proxy Authentication Required
  180. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK48acd271
  181. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  182. t: <sip:4183172685@callcentric.com>
  183. i: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  184. CSeq: 102 INVITE
  185. Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="f8f7301a7548975e316327eb1933ce4b", opaque="", stale=TRUE, algorithm=MD5
  186. l: 0
  187.  
  188. <------------->
  189. --- (8 headers 0 lines) ---
  190. Transmitting (no NAT) to 204.11.192.39:5080:
  191. ACK sip:4183172685@callcentric.com SIP/2.0
  192. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK48acd271
  193. Max-Forwards: 70
  194. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  195. To: <sip:4183172685@callcentric.com>
  196. Contact: <sip:17772409788@142.134.91.178:5060>
  197. Call-ID: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  198. CSeq: 102 ACK
  199. User-Agent: Asterisk PBX 13.6.0
  200. Content-Length: 0
  201.  
  202.  
  203. ---
  204. Audio is at 27464
  205. Adding codec ulaw to SDP
  206. Adding non-codec 0x1 (telephone-event) to SDP
  207. Reliably Transmitting (no NAT) to 204.11.192.39:5080:
  208. INVITE sip:4183172685@callcentric.com SIP/2.0
  209. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK3b4270d6
  210. Max-Forwards: 70
  211. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  212. To: <sip:4183172685@callcentric.com>
  213. Contact: <sip:17772409788@142.134.91.178:5060>
  214. Call-ID: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  215. CSeq: 103 INVITE
  216. User-Agent: Asterisk PBX 13.6.0
  217. Proxy-Authorization: Digest username="17772409788", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="f8f7301a7548975e316327eb1933ce4b", response="602daf81118fccb167a5561306f7b8ec"
  218. Date: Sat, 14 Nov 2015 20:52:01 GMT
  219. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  220. Supported: replaces, timer
  221. Content-Type: application/sdp
  222. Content-Length: 243
  223.  
  224. v=0
  225. o=root 1847464334 1847464335 IN IP4 142.134.91.178
  226. s=Asterisk PBX 13.6.0
  227. c=IN IP4 142.134.91.178
  228. t=0 0
  229. m=audio 27464 RTP/AVP 0 101
  230. a=rtpmap:0 PCMU/8000
  231. a=rtpmap:101 telephone-event/8000
  232. a=fmtp:101 0-16
  233. a=maxptime:150
  234. a=sendrecv
  235.  
  236. ---
  237.  
  238. <--- SIP read from UDP:204.11.192.39:5080 --->
  239. SIP/2.0 403 Incorrect Authentication
  240. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK3b4270d6
  241. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  242. t: <sip:4183172685@callcentric.com>
  243. i: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  244. CSeq: 103 INVITE
  245. l: 0
  246.  
  247. <------------->
  248. --- (7 headers 0 lines) ---
  249. Transmitting (no NAT) to 204.11.192.39:5080:
  250. ACK sip:4183172685@callcentric.com SIP/2.0
  251. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK3b4270d6
  252. Max-Forwards: 70
  253. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as43a72e36
  254. To: <sip:4183172685@callcentric.com>
  255. Contact: <sip:17772409788@142.134.91.178:5060>
  256. Call-ID: 60ad26a764e043a71826037f5e145ebc@callcentric.com
  257. CSeq: 103 ACK
  258. User-Agent: Asterisk PBX 13.6.0
  259. Content-Length: 0
  260.  
  261.  
  262. ---
  263. Scheduling destruction of SIP dialog '60ad26a764e043a71826037f5e145ebc@callcentric.com' in 32000 ms (Method: INVITE)
  264. == Everyone is busy/congested at this time (1:0/0/1)
  265. -- Executing [4183172685@LocalSets:3] PlayTones("SIP/0000FFFF0001-00000000", "congestion") in new stack
  266. -- Executing [4183172685@LocalSets:4] Hangup("SIP/0000FFFF0001-00000000", "") in new stack
  267. == Spawn extension (LocalSets, 4183172685, 4) exited non-zero on 'SIP/0000FFFF0001-00000000'
  268. Scheduling destruction of SIP dialog '1447514424-6108-GAMING-PC@192.168.1.102' in 32000 ms (Method: INVITE)
  269.  
  270. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  271. SIP/2.0 403 Forbidden
  272. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1866108;received=192.168.1.102;rport=5070
  273. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  274. To: <sip:4183172685@192.168.1.207>;tag=as4ee96bc4
  275. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  276. CSeq: 158 INVITE
  277. Server: Asterisk PBX 13.6.0
  278. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  279. Supported: replaces, timer
  280. Content-Length: 0
  281.  
  282.  
  283. <------------>
  284.  
  285. <--- SIP read from UDP:192.168.1.102:5070 --->
  286. ACK sip:4183172685@192.168.1.207 SIP/2.0
  287. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1866108
  288. To: <sip:4183172685@192.168.1.207>;tag=as4ee96bc4
  289. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8335
  290. Call-ID: 1447514424-6108-GAMING-PC@192.168.1.102
  291. CSeq: 158 ACK
  292. Max-Forwards: 20
  293. User-Agent: NCH Software Express Talk 4.35
  294. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="5bc25834",uri="sip:4183172685@192.168.1.207",response="d7fdabc74ad635888637b578b9138b27",opaque="",algorithm=MD5
  295. Content-Length: 0
  296.  
  297. <------------->
  298. --- (10 headers 0 lines) ---
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