Guest User

sipdebug

a guest
Dec 17th, 2015
206
Never
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
  1.  
  2. <--- SIP read from UDP:192.168.1.47:5060 --->
  3. INVITE sip:907611111111@192.168.1.189 SIP/2.0
  4. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;rport
  5. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  6. To: sip:907611111111@192.168.1.189
  7. CSeq: 20 INVITE
  8. Call-ID: 0kYkfVYIit
  9. Max-Forwards: 70
  10. Supported: replaces, outbound
  11. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  12. Content-Type: application/sdp
  13. Content-Length: 465
  14. Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
  15. User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
  16.  
  17. v=0
  18. o=6001 3051 3572 IN IP4 192.168.1.47
  19. s=Talk
  20. c=IN IP4 192.168.1.47
  21. t=0 0
  22. m=audio 7078 RTP/AVP 124 120 111 110 0 8 101
  23. a=rtpmap:124 opus/48000
  24. a=fmtp:124 useinbandfec=1; usedtx=1
  25. a=rtpmap:120 SILK/16000
  26. a=rtpmap:111 speex/16000
  27. a=fmtp:111 vbr=on
  28. a=rtpmap:110 speex/8000
  29. a=fmtp:110 vbr=on
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-15
  32. m=video 9078 RTP/AVP 103 99
  33. a=rtpmap:103 VP8/90000
  34. a=rtpmap:99 MP4V-ES/90000
  35. a=fmtp:99 profile-level-id=3
  36. <------------->
  37. --- (13 headers 19 lines) ---
  38. Sending to 192.168.1.47:5060 (NAT)
  39. Using INVITE request as basis request - 0kYkfVYIit
  40. Found peer '6001' for '6001' from 192.168.1.47:5060
  41.  
  42. <--- Reliably Transmitting (no NAT) to 192.168.1.47:5060 --->
  43. SIP/2.0 401 Unauthorized
  44. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;received=192.168.1.47;rport=5060
  45. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  46. To: sip:907611111111@192.168.1.189;tag=as78e32904
  47. Call-ID: 0kYkfVYIit
  48. CSeq: 20 INVITE
  49. Server: Asterisk PBX 1.8.32.3
  50. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  51. Supported: replaces, timer
  52. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="70289dde"
  53. Content-Length: 0
  54.  
  55.  
  56. <------------>
  57. Scheduling destruction of SIP dialog '0kYkfVYIit' in 32000 ms (Method: INVITE)
  58.  
  59. <--- SIP read from UDP:192.168.1.47:5060 --->
  60. ACK sip:907611111111@192.168.1.189 SIP/2.0
  61. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.YnMq8IiL-;rport
  62. Call-ID: 0kYkfVYIit
  63. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  64. To: <sip:907611111111@192.168.1.189>;tag=as78e32904
  65. Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
  66. Max-Forwards: 70
  67. CSeq: 20 ACK
  68.  
  69. <------------->
  70. --- (8 headers 0 lines) ---
  71.  
  72. <--- SIP read from UDP:192.168.1.47:5060 --->
  73. INVITE sip:907611111111@192.168.1.189 SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
  75. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  76. To: sip:907611111111@192.168.1.189
  77. CSeq: 21 INVITE
  78. Call-ID: 0kYkfVYIit
  79. Max-Forwards: 70
  80. Supported: replaces, outbound
  81. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  82. Content-Type: application/sdp
  83. Content-Length: 465
  84. Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
  85. User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
  86. Authorization: Digest realm="asterisk", nonce="70289dde", username="6001", uri="sip:907611111111@192.168.1.189", response="a3491ceee9e186203855e136a41e7f0f"
  87.  
  88. v=0
  89. o=6001 3051 3572 IN IP4 192.168.1.47
  90. s=Talk
  91. c=IN IP4 192.168.1.47
  92. t=0 0
  93. m=audio 7078 RTP/AVP 124 120 111 110 0 8 101
  94. a=rtpmap:124 opus/48000
  95. a=fmtp:124 useinbandfec=1; usedtx=1
  96. a=rtpmap:120 SILK/16000
  97. a=rtpmap:111 speex/16000
  98. a=fmtp:111 vbr=on
  99. a=rtpmap:110 speex/8000
  100. a=fmtp:110 vbr=on
  101. a=rtpmap:101 telephone-event/8000
  102. a=fmtp:101 0-15
  103. m=video 9078 RTP/AVP 103 99
  104. a=rtpmap:103 VP8/90000
  105. a=rtpmap:99 MP4V-ES/90000
  106. a=fmtp:99 profile-level-id=3
  107. <------------->
  108. --- (14 headers 19 lines) ---
  109. Sending to 192.168.1.47:5060 (no NAT)
  110. Using INVITE request as basis request - 0kYkfVYIit
  111. Found peer '6001' for '6001' from 192.168.1.47:5060
  112. == Using SIP RTP CoS mark 5
  113. Found RTP audio format 124
  114. Found RTP audio format 120
  115. Found RTP audio format 111
  116. Found RTP audio format 110
  117. Found RTP audio format 0
  118. Found RTP audio format 8
  119. Found RTP audio format 101
  120. Found unknown media description format opus for ID 124
  121. Found unknown media description format SILK for ID 120
  122. Found audio description format speex for ID 111
  123. Found audio description format speex for ID 110
  124. Found audio description format telephone-event for ID 101
  125. Found RTP video format 103
  126. Found RTP video format 99
  127. Found video description format MP4V-ES for ID 99
  128. Capabilities: us - 0x4 (ulaw), peer - audio=0x20000020c (ulaw|alaw|speex|speex16)/video=0x500000 (h263p|mpeg4)/text=0x0 (nothing), combined - 0x4 (ulaw)
  129. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  130. Peer audio RTP is at port 192.168.1.47:7078
  131. Looking for 907611111111 in office (domain 192.168.1.189)
  132. list_route: hop: <sip:6001@192.168.1.47>
  133.  
  134. <--- Transmitting (no NAT) to 192.168.1.47:5060 --->
  135. SIP/2.0 100 Trying
  136. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
  137. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  138. To: sip:907611111111@192.168.1.189
  139. Call-ID: 0kYkfVYIit
  140. CSeq: 21 INVITE
  141. Server: Asterisk PBX 1.8.32.3
  142. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  143. Supported: replaces, timer
  144. Contact: <sip:907611111111@192.168.1.189:5060>
  145. Content-Length: 0
  146.  
  147.  
  148. ------------>
  149. -- Executing [907611111111@office:1] Log("SIP/6001-00000008", "NOTICE, Dialing out from "" <6001> to 907611111111 via HT503") in new stack
  150. [Dec 17 09:01:50] NOTICE[1344]: Ext. 907611111111:1 @ office: Dialing out from "" <6001> to 907611111111 via HT503
  151. -- Executing [907611111111@office:2] MixMonitor("SIP/6001-00000008", "1450342910-office-907611111111.wav") in new stack
  152. -- Executing [907611111111@office:3] Set("SIP/6001-00000008", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
  153. -- Executing [907611111111@office:4] Dial("SIP/6001-00000008", "SIP/ht503fxo/907611111111,60") in new stack
  154. == Using SIP RTP CoS mark 5
  155. Audio is at 18872
  156. Adding codec 0x4 (ulaw) to SDP
  157. Adding codec 0x800 (g726) to SDP
  158. Adding codec 0x2 (gsm) to SDP
  159. Adding non-codec 0x1 (telephone-event) to SDP
  160. Reliably Transmitting (no NAT) to 192.168.1.120:5062:
  161. INVITE sip:907611111111@192.168.1.120 SIP/2.0
  162. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  163. Max-Forwards: 70
  164. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  165. To: <sip:907611111111@192.168.1.120>
  166. Contact: <sip:6001@192.168.1.189:5060>
  167. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  168. CSeq: 102 INVITE
  169. User-Agent: Asterisk PBX 1.8.32.3
  170. Date: Thu, 17 Dec 2015 09:01:50 GMT
  171. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  172. Supported: replaces, timer
  173. Content-Type: application/sdp
  174. Content-Length: 291
  175.  
  176. v=0
  177. o=root 523463639 523463639 IN IP4 192.168.1.189
  178. s=Asterisk PBX 1.8.32.3
  179. c=IN IP4 192.168.1.189
  180. t=0 0
  181. m=audio 18872 RTP/AVP 0 111 3 101
  182. a=rtpmap:0 PCMU/8000
  183. a=rtpmap:111 G726-32/8000
  184. a=rtpmap:3 GSM/8000
  185. a=rtpmap:101 telephone-event/8000
  186. a=fmtp:101 0-16
  187. a=ptime:20
  188. a=sendrecv
  189.  
  190. ---
  191. -- Called SIP/ht503fxo/907611111111
  192. == Begin MixMonitor Recording SIP/6001-00000008
  193.  
  194. <--- SIP read from UDP:192.168.1.120:5062 --->
  195. SIP/2.0 100 Trying
  196. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  197. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  198. To: <sip:907611111111@192.168.1.120>
  199. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  200. CSeq: 102 INVITE
  201. Supported: replaces, path, timer, eventlist
  202. User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
  203. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  204. Content-Length: 0
  205.  
  206. <------------->
  207. --- (10 headers 0 lines) ---
  208.  
  209. <--- SIP read from UDP:192.168.1.47:5060 --->
  210.  
  211.  
  212. <------------->
  213.  
  214. <--- SIP read from UDP:192.168.1.47:5060 --->
  215.  
  216.  
  217. <------------->
  218.  
  219. <--- SIP read from UDP:192.168.1.47:5060 --->
  220.  
  221.  
  222. <------------->
  223.  
  224. <--- SIP read from UDP:192.168.1.47:5060 --->
  225. CANCEL sip:907611111111@192.168.1.189 SIP/2.0
  226. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
  227. Call-ID: 0kYkfVYIit
  228. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  229. To: sip:907611111111@192.168.1.189
  230. Max-Forwards: 70
  231. CSeq: 21 CANCEL
  232. User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
  233.  
  234. <------------->
  235. --- (8 headers 0 lines) ---
  236. Sending to 192.168.1.47:5060 (no NAT)
  237.  
  238. <--- Reliably Transmitting (no NAT) to 192.168.1.47:5060 --->
  239. SIP/2.0 487 Request Terminated
  240. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
  241. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  242. To: sip:907611111111@192.168.1.189;tag=as17389654
  243. Call-ID: 0kYkfVYIit
  244. CSeq: 21 INVITE
  245. Server: Asterisk PBX 1.8.32.3
  246. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  247. Supported: replaces, timer
  248. Content-Length: 0
  249.  
  250.  
  251. <------------>
  252.  
  253. <--- Transmitting (no NAT) to 192.168.1.47:5060 --->
  254. SIP/2.0 200 OK
  255. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;received=192.168.1.47;rport=5060
  256. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  257. To: sip:907611111111@192.168.1.189;tag=as17389654
  258. Call-ID: 0kYkfVYIit
  259. CSeq: 21 CANCEL
  260. Server: Asterisk PBX 1.8.32.3
  261. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  262. Supported: replaces, timer
  263. Content-Length: 0
  264.  
  265.  
  266. <------------>
  267. Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
  268. Reliably Transmitting (no NAT) to 192.168.1.120:5062:
  269. CANCEL sip:907611111111@192.168.1.120 SIP/2.0
  270. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  271. Max-Forwards: 70
  272. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  273. To: <sip:907611111111@192.168.1.120>
  274. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  275. CSeq: 102 CANCEL
  276. User-Agent: Asterisk PBX 1.8.32.3
  277. Content-Length: 0
  278.  
  279.  
  280. ---
  281. Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
  282. == Spawn extension (office, 907611111111, 4) exited non-zero on 'SIP/6001-00000008'
  283. == End MixMonitor Recording SIP/6001-00000008
  284.  
  285. <--- SIP read from UDP:192.168.1.47:5060 --->
  286. ACK sip:907611111111@192.168.1.189 SIP/2.0
  287. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK.un3g8uiDk;rport
  288. Call-ID: 0kYkfVYIit
  289. From: <sip:6001@192.168.1.189>;tag=RaXJH616h
  290. To: <sip:907611111111@192.168.1.189>;tag=as17389654
  291. Contact: <sip:6001@192.168.1.47>;+sip.instance="<urn:uuid:23f4425c-4b3e-409c-a2aa-f339203fffbb>"
  292. Max-Forwards: 70
  293. CSeq: 21 ACK
  294.  
  295. <------------->
  296. --- (8 headers 0 lines) ---
  297. Really destroying SIP dialog '0kYkfVYIit' Method: ACK
  298.  
  299. <--- SIP read from UDP:192.168.1.120:5062 --->
  300. SIP/2.0 487 Request Terminated
  301. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  302. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  303. To: <sip:907611111111@192.168.1.120>;tag=568004079
  304. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  305. CSeq: 102 INVITE
  306. Supported: replaces, path, timer, eventlist
  307. User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
  308. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  309. Content-Length: 0
  310.  
  311. <------------->
  312. --- (10 headers 0 lines) ---
  313. Transmitting (no NAT) to 192.168.1.120:5062:
  314. ACK sip:907611111111@192.168.1.120 SIP/2.0
  315. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  316. Max-Forwards: 70
  317. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  318. To: <sip:907611111111@192.168.1.120>;tag=568004079
  319. Contact: <sip:6001@192.168.1.189:5060>
  320. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  321. CSeq: 102 ACK
  322. User-Agent: Asterisk PBX 1.8.32.3
  323. Content-Length: 0
  324.  
  325.  
  326. ---
  327. Scheduling destruction of SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' in 32000 ms (Method: INVITE)
  328.  
  329. <--- SIP read from UDP:192.168.1.120:5062 --->
  330. SIP/2.0 200 OK
  331. Via: SIP/2.0/UDP 192.168.1.189:5060;branch=z9hG4bK69d6ce7e
  332. From: "6001" <sip:6001@192.168.1.189>;tag=as0687b888
  333. To: <sip:907611111111@192.168.1.120>;tag=568004079
  334. Call-ID: 7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060
  335. CSeq: 102 CANCEL
  336. Contact: <sip:ht503fxo@192.168.1.120:5062>
  337. Supported: replaces, path, timer, eventlist
  338. User-Agent: Grandstream HT-503 V2.0A 1.0.12.1 chip V2.2
  339. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  340. Content-Length: 0
  341.  
  342. <------------->
  343. --- (11 headers 0 lines) ---
  344.  
  345. <--- SIP read from UDP:192.168.1.47:5060 --->
  346.  
  347.  
  348. <------------->
  349.  
  350. <--- SIP read from UDP:192.168.1.47:5060 --->
  351.  
  352.  
  353. <------------->
  354.  
  355. <--- SIP read from UDP:192.168.1.47:5060 --->
  356.  
  357.  
  358. <------------->
  359. Really destroying SIP dialog '7354987b45a3028200417b7d4e0d4241@192.168.1.189:5060' Method: INVITE
RAW Paste Data