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  1. <--- SIP read from UDP:192.168.0.6:5062 --->
  2. INVITE sip:incoming@192.168.0.2:5060 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK2106295293;rport
  4. From: <sip:0989757267@192.168.0.2>;tag=730509171
  5. To: <sip:incoming@192.168.0.2:5060>
  6. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  7. CSeq: 50 INVITE
  8. Contact: <sip:FXO@192.168.0.6:5062>
  9. Max-Forwards: 70
  10. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  11. Supported: replaces, path, timer, eventlist
  12. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  13. Content-Type: application/sdp
  14. Accept: application/sdp, application/dtmf-relay
  15. Content-Length: 384
  16.  
  17. v=0
  18. o=FXO 8002 8000 IN IP4 192.168.0.6
  19. s=SIP Call
  20. c=IN IP4 192.168.0.6
  21. t=0 0
  22. m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100
  23. a=sendrecv
  24. a=rtpmap:0 PCMU/8000
  25. a=ptime:20
  26. a=rtpmap:8 PCMA/8000
  27. a=rtpmap:4 G723/8000
  28. a=rtpmap:18 G729/8000
  29. a=fmtp:18 annexb=no
  30. a=rtpmap:2 G726-32/8000
  31. a=rtpmap:97 iLBC/8000
  32. a=fmtp:97 mode=20
  33. a=rtpmap:102 G729E/8000
  34. a=rtpmap:100 AAL2-G726-16/8000
  35. <------------->
  36. --- (14 headers 18 lines) ---
  37. Sending to 192.168.0.6:5062 (no NAT)
  38. Sending to 192.168.0.6:5062 (no NAT)
  39. Using INVITE request as basis request - 2125658203-5062-6@BJC.BGI.A.G
  40. Found peer 'FXO' for '0989757267' from 192.168.0.6:5062
  41.  
  42. <--- Reliably Transmitting (no NAT) to 192.168.0.6:5062 --->
  43. SIP/2.0 401 Unauthorized
  44. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK2106295293;received=192.168.0.6;rport=5062
  45. From: <sip:0989757267@192.168.0.2>;tag=730509171
  46. To: <sip:incoming@192.168.0.2:5060>;tag=as5f2adfca
  47. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  48. CSeq: 50 INVITE
  49. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  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="38e17c6b"
  53. Content-Length: 0
  54.  
  55.  
  56. <------------>
  57. Scheduling destruction of SIP dialog '2125658203-5062-6@BJC.BGI.A.G' in 6400 ms (Method: INVITE)
  58.  
  59. <--- SIP read from UDP:192.168.0.6:5062 --->
  60. ACK sip:incoming@192.168.0.2:5060 SIP/2.0
  61. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK2106295293;rport
  62. From: <sip:0989757267@192.168.0.2>;tag=730509171
  63. To: <sip:incoming@192.168.0.2:5060>;tag=as5f2adfca
  64. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  65. CSeq: 50 ACK
  66. Content-Length: 0
  67.  
  68. <------------->
  69. --- (7 headers 0 lines) ---
  70.  
  71. <--- SIP read from UDP:192.168.0.6:5062 --->
  72. INVITE sip:incoming@192.168.0.2:5060 SIP/2.0
  73. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1202600157;rport
  74. From: <sip:0989757267@192.168.0.2>;tag=730509171
  75. To: <sip:incoming@192.168.0.2:5060>
  76. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  77. CSeq: 51 INVITE
  78. Contact: <sip:FXO@192.168.0.6:5062>
  79. Authorization: Digest username="FXO", realm="asterisk", nonce="38e17c6b", uri="sip:incoming@192.168.0.2:5060", response="86843b4e79fd893b816a6486e2f759f7", algorithm=MD5
  80. Max-Forwards: 70
  81. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  82. Supported: replaces, path, timer, eventlist
  83. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  84. Content-Type: application/sdp
  85. Accept: application/sdp, application/dtmf-relay
  86. Content-Length: 384
  87.  
  88. v=0
  89. o=FXO 8002 8000 IN IP4 192.168.0.6
  90. s=SIP Call
  91. c=IN IP4 192.168.0.6
  92. t=0 0
  93. m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100
  94. a=sendrecv
  95. a=rtpmap:0 PCMU/8000
  96. a=ptime:20
  97. a=rtpmap:8 PCMA/8000
  98. a=rtpmap:4 G723/8000
  99. a=rtpmap:18 G729/8000
  100. a=fmtp:18 annexb=no
  101. a=rtpmap:2 G726-32/8000
  102. a=rtpmap:97 iLBC/8000
  103. a=fmtp:97 mode=20
  104. a=rtpmap:102 G729E/8000
  105. a=rtpmap:100 AAL2-G726-16/8000
  106. <------------->
  107. --- (15 headers 18 lines) ---
  108. Sending to 192.168.0.6:5062 (no NAT)
  109. Using INVITE request as basis request - 2125658203-5062-6@BJC.BGI.A.G
  110. Found peer 'FXO' for '0989757267' from 192.168.0.6:5062
  111. == Using SIP RTP CoS mark 5
  112. Found RTP audio format 0
  113. Found RTP audio format 8
  114. Found RTP audio format 4
  115. Found RTP audio format 18
  116. Found RTP audio format 2
  117. Found RTP audio format 97
  118. Found RTP audio format 102
  119. Found RTP audio format 100
  120. Found audio description format PCMU for ID 0
  121. Found audio description format PCMA for ID 8
  122. Found audio description format G723 for ID 4
  123. Found audio description format G729 for ID 18
  124. Found audio description format G726-32 for ID 2
  125. Found audio description format iLBC for ID 97
  126. Found unknown media description format G729E for ID 102
  127. Found unknown media description format AAL2-G726-16 for ID 100
  128. Capabilities: us - (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
  129. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  130. Peer audio RTP is at port 192.168.0.6:5013
  131. Looking for incoming in phones (domain 192.168.0.2)
  132. list_route: hop: <sip:FXO@192.168.0.6:5062>
  133.  
  134. <--- Transmitting (no NAT) to 192.168.0.6:5062 --->
  135. SIP/2.0 100 Trying
  136. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1202600157;received=192.168.0.6;rport=5062
  137. From: <sip:0989757267@192.168.0.2>;tag=730509171
  138. To: <sip:incoming@192.168.0.2:5060>
  139. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  140. CSeq: 51 INVITE
  141. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  142. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  143. Supported: replaces, timer
  144. Session-Expires: 1800;refresher=uas
  145. Contact: <sip:incoming@192.168.0.2:5060>
  146. Content-Length: 0
  147.  
  148.  
  149. <------------>
  150. -- Executing [incoming@phones:1] GotoIfTime("SIP/FXO-0000001a", "08:00-21:59,sun-fri,*,*?open:closed") in new stack
  151. -- Goto (phones,incoming,5)
  152. -- Executing [incoming@phones:5] Set("SIP/FXO-0000001a", "VOLUME(TX)=25") in new stack
  153. -- Executing [incoming@phones:6] Answer("SIP/FXO-0000001a", "") in new stack
  154. Audio is at 14284
  155. Adding codec 100003 (ulaw) to SDP
  156. Adding codec 100004 (alaw) to SDP
  157. Adding codec 100008 (g729) to SDP
  158.  
  159. <--- Reliably Transmitting (no NAT) to 192.168.0.6:5062 --->
  160. SIP/2.0 200 OK
  161. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1202600157;received=192.168.0.6;rport=5062
  162. From: <sip:0989757267@192.168.0.2>;tag=730509171
  163. To: <sip:incoming@192.168.0.2:5060>;tag=as58e36e6b
  164. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  165. CSeq: 51 INVITE
  166. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  168. Supported: replaces, timer
  169. Session-Expires: 1800;refresher=uas
  170. Contact: <sip:incoming@192.168.0.2:5060>
  171. Content-Type: application/sdp
  172. Require: timer
  173. Content-Length: 263
  174.  
  175. v=0
  176. o=root 1479357588 1479357588 IN IP4 192.168.0.2
  177. s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
  178. c=IN IP4 192.168.0.2
  179. t=0 0
  180. m=audio 14284 RTP/AVP 0 8 18
  181. a=rtpmap:0 PCMU/8000
  182. a=rtpmap:8 PCMA/8000
  183. a=rtpmap:18 G729/8000
  184. a=fmtp:18 annexb=no
  185. a=ptime:20
  186. a=sendrecv
  187.  
  188. <------------>
  189.  
  190. <--- SIP read from UDP:192.168.0.6:5062 --->
  191. ACK sip:incoming@192.168.0.2:5060 SIP/2.0
  192. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1828988367;rport
  193. From: <sip:0989757267@192.168.0.2>;tag=730509171
  194. To: <sip:incoming@192.168.0.2:5060>;tag=as58e36e6b
  195. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  196. CSeq: 51 ACK
  197. Contact: <sip:FXO@192.168.0.6:5062>
  198. Max-Forwards: 70
  199. Supported: replaces, path, timer, eventlist
  200. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  201. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  202. Content-Length: 0
  203.  
  204. <------------->
  205. --- (12 headers 0 lines) ---
  206. > 0x7f3cc4029870 -- Probation passed - setting RTP source address to 192.168.0.6:5013
  207. -- Executing [incoming@phones:7] Dial("SIP/FXO-0000001a", "SIP/101,897,m(default)") in new stack
  208. == Using SIP RTP CoS mark 5
  209. Audio is at 13824
  210. Adding codec 100003 (ulaw) to SDP
  211. Adding codec 100004 (alaw) to SDP
  212. Adding codec 100002 (gsm) to SDP
  213. Adding codec 100017 (testlaw) to SDP
  214. Adding non-codec 0x1 (telephone-event) to SDP
  215. Reliably Transmitting (no NAT) to 192.168.0.101:5060:
  216. INVITE sip:101@192.168.0.101:5060 SIP/2.0
  217. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d04b54b
  218. Max-Forwards: 70
  219. From: <sip:0989757267@192.168.0.2>;tag=as178183e7
  220. To: <sip:101@192.168.0.101:5060>
  221. Contact: <sip:0989757267@192.168.0.2:5060>
  222. Call-ID: 0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060
  223. CSeq: 102 INVITE
  224. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  225. Date: Tue, 14 Mar 2017 09:57:44 GMT
  226. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  227. Supported: replaces, timer
  228. Content-Type: application/sdp
  229. Content-Length: 295
  230.  
  231. v=0
  232. o=root 2052080509 2052080509 IN IP4 192.168.0.2
  233. s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
  234. c=IN IP4 192.168.0.2
  235. t=0 0
  236. m=audio 13824 RTP/AVP 0 8 3 101
  237. a=rtpmap:0 PCMU/8000
  238. a=rtpmap:8 PCMA/8000
  239. a=rtpmap:3 GSM/8000
  240. a=rtpmap:101 telephone-event/8000
  241. a=fmtp:101 0-16
  242. a=ptime:20
  243. a=sendrecv
  244.  
  245. ---
  246. -- Called SIP/101
  247. -- Started music on hold, class 'default', on SIP/FXO-0000001a
  248.  
  249. <--- SIP read from UDP:192.168.0.101:5060 --->
  250. SIP/2.0 100 Trying
  251. To: <sip:101@192.168.0.101:5060>
  252. From: <sip:0989757267@192.168.0.2>;tag=as178183e7
  253. Call-ID: 0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060
  254. CSeq: 102 INVITE
  255. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d04b54b
  256. Server: Linksys/SPA942-6.1.5(a)
  257. Content-Length: 0
  258.  
  259. <------------->
  260. --- (8 headers 0 lines) ---
  261. [Mar 14 10:57:44] WARNING[26198][C-0000000d]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304
  262.  
  263. <--- SIP read from UDP:192.168.0.101:5060 --->
  264. SIP/2.0 180 Ringing
  265. To: <sip:101@192.168.0.101:5060>;tag=7120f913269512di0
  266. From: <sip:0989757267@192.168.0.2>;tag=as178183e7
  267. Call-ID: 0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060
  268. CSeq: 102 INVITE
  269. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d04b54b
  270. Contact: "Vlatko" <sip:101@192.168.0.101:5060>
  271. Server: Linksys/SPA942-6.1.5(a)
  272. Content-Length: 0
  273.  
  274. <------------->
  275. --- (9 headers 0 lines) ---
  276. list_route: hop: <sip:101@192.168.0.101:5060>
  277. -- SIP/101-0000001b is ringing
  278. Reliably Transmitting (no NAT) to 192.168.0.6:5062:
  279. OPTIONS sip:FXO@192.168.0.6:5062 SIP/2.0
  280. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3172cd08
  281. Max-Forwards: 70
  282. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as07a97c33
  283. To: <sip:FXO@192.168.0.6:5062>
  284. Contact: <sip:asterisk@192.168.0.2:5060>
  285. Call-ID: 181ad78a09647ac559869d07718f12df@192.168.0.2:5060
  286. CSeq: 102 OPTIONS
  287. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  288. Date: Tue, 14 Mar 2017 09:57:57 GMT
  289. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  290. Supported: replaces, timer
  291. Content-Length: 0
  292.  
  293.  
  294. ---
  295.  
  296. <--- SIP read from UDP:192.168.0.6:5062 --->
  297. SIP/2.0 200 OK
  298. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3172cd08
  299. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as07a97c33
  300. To: <sip:FXO@192.168.0.6:5062>;tag=244095267
  301. Call-ID: 181ad78a09647ac559869d07718f12df@192.168.0.2:5060
  302. CSeq: 102 OPTIONS
  303. Supported: replaces, path, timer, eventlist
  304. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  305. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  306. Content-Length: 0
  307.  
  308. <------------->
  309. --- (10 headers 0 lines) ---
  310. Really destroying SIP dialog '181ad78a09647ac559869d07718f12df@192.168.0.2:5060' Method: OPTIONS
  311. Reliably Transmitting (no NAT) to 192.168.0.4:47151:
  312. OPTIONS sip:105@192.168.0.4:47151;app-id=929724111839;pn-type=firebase;pn-tok=APA91bGtsN_pJHUxAOTJUuVXfzbx7Hgg2710-oGRIB7dbOLpT2akBNYKKXfq-WXT6JYl-hQM6E2eSjXspWDZY632x8Bt4P76nbw3jM_zAoF1qPUqSda8VQI;transport=udp SIP/2.0
  313. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK68559307
  314. Max-Forwards: 70
  315. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as4b767b52
  316. To: <sip:105@192.168.0.4:47151;app-id=929724111839;pn-type=firebase;pn-tok=APA91bGtsN_pJHUxAOTJUuVXfzbx7Hgg2710-oGRIB7dbOLpT2akBNYKKXfq-WXT6JYl-hQM6E2eSjXspWDZY632x8Bt4P76nbw3jM_zAoF1qPUqSda8VQI;transport=udp>
  317. Contact: <sip:asterisk@192.168.0.2:5060>
  318. Call-ID: 5b9829e2178d8c1803eb6f0312ed84c9@192.168.0.2:5060
  319. CSeq: 102 OPTIONS
  320. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  321. Date: Tue, 14 Mar 2017 09:58:26 GMT
  322. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  323. Supported: replaces, timer
  324. Content-Length: 0
  325.  
  326.  
  327. ---
  328.  
  329. <--- SIP read from UDP:192.168.0.4:47151 --->
  330. SIP/2.0 200 Ok
  331. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK68559307
  332. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as4b767b52
  333. To: <sip:105@192.168.0.4:47151;app-id=929724111839;pn-type=firebase;pn-tok=APA91bGtsN_pJHUxAOTJUuVXfzbx7Hgg2710-oGRIB7dbOLpT2akBNYKKXfq-WXT6JYl-hQM6E2eSjXspWDZY632x8Bt4P76nbw3jM_zAoF1qPUqSda8VQI;transport=udp>;tag=zuE~N
  334. Call-ID: 5b9829e2178d8c1803eb6f0312ed84c9@192.168.0.2:5060
  335. CSeq: 102 OPTIONS
  336.  
  337. <------------->
  338. --- (6 headers 0 lines) ---
  339. Really destroying SIP dialog '5b9829e2178d8c1803eb6f0312ed84c9@192.168.0.2:5060' Method: OPTIONS
  340. Reliably Transmitting (no NAT) to 192.168.0.101:5060:
  341. OPTIONS sip:101@192.168.0.101:5060 SIP/2.0
  342. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK5a15e131
  343. Max-Forwards: 70
  344. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as6b66380c
  345. To: <sip:101@192.168.0.101:5060>
  346. Contact: <sip:asterisk@192.168.0.2:5060>
  347. Call-ID: 5f4daa153275f6355fa532df6b8af9b8@192.168.0.2:5060
  348. CSeq: 102 OPTIONS
  349. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  350. Date: Tue, 14 Mar 2017 09:58:27 GMT
  351. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  352. Supported: replaces, timer
  353. Content-Length: 0
  354.  
  355.  
  356. ---
  357.  
  358. <--- SIP read from UDP:192.168.0.101:5060 --->
  359. SIP/2.0 200 OK
  360. To: <sip:101@192.168.0.101:5060>;tag=fbbabdbd9ff0cd3fi0
  361. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as6b66380c
  362. Call-ID: 5f4daa153275f6355fa532df6b8af9b8@192.168.0.2:5060
  363. CSeq: 102 OPTIONS
  364. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK5a15e131
  365. Server: Linksys/SPA942-6.1.5(a)
  366. Content-Length: 0
  367. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  368. Supported: replaces
  369.  
  370. <------------->
  371. --- (10 headers 0 lines) ---
  372. Really destroying SIP dialog '5f4daa153275f6355fa532df6b8af9b8@192.168.0.2:5060' Method: OPTIONS
  373. Reliably Transmitting (no NAT) to 192.168.0.6:5060:
  374. OPTIONS sip:FXS@192.168.0.6:5060 SIP/2.0
  375. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK75aaa56f
  376. Max-Forwards: 70
  377. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as3748c0a1
  378. To: <sip:FXS@192.168.0.6:5060>
  379. Contact: <sip:asterisk@192.168.0.2:5060>
  380. Call-ID: 55c611b72b92cd9b37a65e783cbbdc7d@192.168.0.2:5060
  381. CSeq: 102 OPTIONS
  382. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  383. Date: Tue, 14 Mar 2017 09:58:43 GMT
  384. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  385. Supported: replaces, timer
  386. Content-Length: 0
  387.  
  388.  
  389. ---
  390.  
  391. <--- SIP read from UDP:192.168.0.6:5060 --->
  392. SIP/2.0 200 OK
  393. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK75aaa56f
  394. From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as3748c0a1
  395. To: <sip:FXS@192.168.0.6:5060>;tag=249064437
  396. Call-ID: 55c611b72b92cd9b37a65e783cbbdc7d@192.168.0.2:5060
  397. CSeq: 102 OPTIONS
  398. Supported: replaces, path, timer, eventlist
  399. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  400. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  401. Content-Length: 0
  402.  
  403. <------------->
  404. --- (10 headers 0 lines) ---
  405. Really destroying SIP dialog '55c611b72b92cd9b37a65e783cbbdc7d@192.168.0.2:5060' Method: OPTIONS
  406.  
  407. <--- SIP read from UDP:192.168.0.101:5060 --->
  408. SIP/2.0 480 Temporarily not available
  409. To: <sip:101@192.168.0.101:5060>;tag=7120f913269512di0
  410. From: <sip:0989757267@192.168.0.2>;tag=as178183e7
  411. Call-ID: 0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060
  412. CSeq: 102 INVITE
  413. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d04b54b
  414. Server: Linksys/SPA942-6.1.5(a)
  415. Content-Length: 0
  416.  
  417. <------------->
  418. --- (8 headers 0 lines) ---
  419. -- Got SIP response 480 "Temporarily not available" back from 192.168.0.101:5060
  420. set_destination: Parsing <sip:101@192.168.0.101:5060> for address/port to send to
  421. set_destination: set destination to 192.168.0.101:5060
  422. Transmitting (no NAT) to 192.168.0.101:5060:
  423. ACK sip:101@192.168.0.101:5060 SIP/2.0
  424. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6d04b54b
  425. Max-Forwards: 70
  426. From: <sip:0989757267@192.168.0.2>;tag=as178183e7
  427. To: <sip:101@192.168.0.101:5060>;tag=7120f913269512di0
  428. Contact: <sip:0989757267@192.168.0.2:5060>
  429. Call-ID: 0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060
  430. CSeq: 102 ACK
  431. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  432. Content-Length: 0
  433.  
  434.  
  435. ---
  436. -- SIP/101-0000001b is circuit-busy
  437. == Everyone is busy/congested at this time (1:0/1/0)
  438. -- Stopped music on hold on SIP/FXO-0000001a
  439. -- Executing [incoming@phones:8] Hangup("SIP/FXO-0000001a", "") in new stack
  440. == Spawn extension (phones, incoming, 8) exited non-zero on 'SIP/FXO-0000001a'
  441. Scheduling destruction of SIP dialog '2125658203-5062-6@BJC.BGI.A.G' in 6400 ms (Method: ACK)
  442. set_destination: Parsing <sip:FXO@192.168.0.6:5062> for address/port to send to
  443. set_destination: set destination to 192.168.0.6:5062
  444. Reliably Transmitting (no NAT) to 192.168.0.6:5062:
  445. BYE sip:FXO@192.168.0.6:5062 SIP/2.0
  446. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3a4bcde2;rport
  447. Max-Forwards: 70
  448. From: <sip:incoming@192.168.0.2:5060>;tag=as58e36e6b
  449. To: <sip:0989757267@192.168.0.2>;tag=730509171
  450. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  451. CSeq: 102 BYE
  452. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
  453. Proxy-Authorization: Digest username="FXO", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.2", nonce="38e17c6b", response="7271886424675cd8026e0c505c275968"
  454. X-Asterisk-HangupCause: User alerting, no answer
  455. X-Asterisk-HangupCauseCode: 19
  456. Content-Length: 0
  457.  
  458.  
  459. ---
  460. Really destroying SIP dialog '0c4a1f4c7d9e21b947b6c93537c38a83@192.168.0.2:5060' Method: INVITE
  461.  
  462. <--- SIP read from UDP:192.168.0.6:5062 --->
  463. SIP/2.0 200 OK
  464. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK3a4bcde2;rport=5060
  465. From: <sip:incoming@192.168.0.2:5060>;tag=as58e36e6b
  466. To: <sip:0989757267@192.168.0.2>;tag=730509171
  467. Call-ID: 2125658203-5062-6@BJC.BGI.A.G
  468. CSeq: 102 BYE
  469. Contact: <sip:FXO@192.168.0.6:5062>
  470. Supported: replaces, path, timer, eventlist
  471. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  472. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  473. Content-Length: 0
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