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  1. sip.conf
  2. [general]
  3. context=default
  4. dtmfmode=rfc2833
  5. relaxdtmf=yes
  6. rfc2833compensate=yest38pt_udptl = yes
  7.  
  8. ; Register with provider for incoming calls
  9. register=>username:password@sip.nettel.dk/my-number
  10.  
  11.  
  12. [my-number]
  13. type=friend
  14. secret=xxxxxx
  15. defaultuser=xxxxxx
  16. host=sip.nettel.dk
  17. fromuser=xxxxxxx
  18. fromdomain=nettel.dk
  19. context=incoming
  20. insecure=invite,port
  21. dtmfmode=rfc2833
  22. qualify=yes
  23. directmedia=no
  24. t38pt_udptl=yes,redundancy
  25. disallow=all
  26. allow=alaw
  27. allow=ulaw
  28.  
  29.  
  30. Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  31.  
  32. *CLI> fax show settings
  33. FAX For Asterisk Settings:
  34. ECM: Enabled
  35. Status Events: On
  36. Minimum Bit Rate: 2400
  37. Maximum Bit Rate: 14400
  38. Modem Modulations Allowed: V17,V27,V29
  39. T.38 Negotiation Timeout: 5000
  40.  
  41.  
  42. FAX Technology Modules:
  43.  
  44. Spandsp (Spandsp FAX Driver) Settings:
  45.  
  46. *CLI> sip show peer my-number
  47.  
  48. * Name : my-number
  49. Description :
  50. Secret : <Set>
  51. MD5Secret : <Not set>
  52. Remote Secret: <Not set>
  53. Context : incoming
  54. Record On feature : automon
  55. Record Off feature : automon
  56. Subscr.Cont. : <Not set>
  57. Language :
  58. Tonezone : <Not set>
  59. AMA flags : Unknown
  60. Transfer mode: open
  61. CallingPres : Presentation Allowed, Not Screened
  62. FromUser : xxxxxxxx
  63. FromDomain : nettel.dk Port 5060
  64. Callgroup :
  65. Pickupgroup :
  66. Named Callgr :
  67. Nam. Pickupgr:
  68. MOH Suggest :
  69. Mailbox :
  70. VM Extension : asterisk
  71. LastMsgsSent : 0/0
  72. Call limit : 0
  73. Max forwards : 0
  74. Dynamic : No
  75. Callerid : "" <>
  76. MaxCallBR : 384 kbps
  77. Expire : -1
  78. Insecure : port,invite
  79. Force rport : Auto (No)
  80. Symmetric RTP: No
  81. ACL : No
  82. DirectMedACL : No
  83. T.38 support : Yes
  84. T.38 EC mode : Redundancy
  85. T.38 MaxDtgrm: 4294967295
  86. DirectMedia : No
  87. PromiscRedir : No
  88. User=Phone : No
  89. Video Support: No
  90. Text Support : No
  91. Ign SDP ver : No
  92. Trust RPID : No
  93. Send RPID : No
  94. Path support : No
  95. Path : N/A
  96. TrustIDOutbnd: Legacy
  97. Subscriptions: Yes
  98. Overlap dial : Yes
  99. DTMFmode : rfc2833
  100. Timer T1 : 500
  101. Timer B : 32000
  102. ToHost : sip.nettel.dk
  103. Addr->IP : 194.192.15.56:5060
  104. Defaddr->IP : (null)
  105. Prim.Transp. : UDP
  106. Allowed.Trsp : UDP
  107. Def. Username: xxxxxxxx
  108. SIP Options : (none)
  109. Codecs : (alaw|ulaw)
  110. Auto-Framing : No
  111. Status : OK (113 ms)
  112. Useragent :
  113. Reg. Contact :
  114. Qualify Freq : 60000 ms
  115. Keepalive : 0 ms
  116. Sess-Timers : Accept
  117. Sess-Refresh : uas
  118. Sess-Expires : 1800 secs
  119. Min-Sess : 90 secs
  120. RTP Engine : asterisk
  121. Parkinglot :
  122. Use Reason : No
  123. Encryption : No
  124.  
  125.  
  126. Trace of incoming FAX:
  127.  
  128. <--- SIP read from UDP:194.192.15.56:5060 --->
  129. INVITE sip:my-number@my-ip-address:5060 SIP/2.0
  130. CSeq: 1 INVITE
  131. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995421
  132. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  133. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  134. To: <sip:my-number@my-ip-address:5060>
  135. Contact: <sip:194.192.15.56:5060;transport=udp>
  136. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
  137. Content-Type: application/sdp
  138. Content-Length: 209
  139. Max-Forwards: 70
  140.  
  141. v=0
  142. o=- 2289845697 2224995406 IN IP4 194.192.15.56
  143. s=VoipSIP
  144. c=IN IP4 194.192.15.56
  145. t=0 0
  146. m=audio 6418 RTP/AVP 8 101
  147. a=rtpmap:8 PCMA/8000
  148. a=rtpmap:101 telephone-event/8000
  149. a=fmtp:101 0-15
  150. a=sendrecv
  151. <------------->
  152. --- (11 headers 10 lines) ---
  153. Sending to 194.192.15.56:5060 (no NAT)
  154. Sending to 194.192.15.56:5060 (no NAT)
  155. Using INVITE request as basis request - 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  156. Found peer 'my-number' for '+41225185576' from 194.192.15.56:5060
  157. == Using SIP RTP CoS mark 5
  158. Found RTP audio format 8
  159. Found RTP audio format 101
  160. Found audio description format PCMA for ID 8
  161. Found audio description format telephone-event for ID 101
  162. Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  163. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  164. Peer audio RTP is at port 194.192.15.56:6418
  165. Looking for my-number in incoming (domain my-ip-address)
  166. sip_route_dump: route/path hop: <sip:194.192.15.56:5060;transport=udp>
  167.  
  168. <--- Transmitting (no NAT) to 194.192.15.56:5060 --->
  169. SIP/2.0 100 Trying
  170. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995421;received=194.192.15.56
  171. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  172. To: <sip:my-number@my-ip-address:5060>
  173. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  174. CSeq: 1 INVITE
  175. Server: Asterisk PBX 13.13.1
  176. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  177. Supported: replaces, timer
  178. Contact: <sip:my-number@my-ip-address:5060>
  179. Content-Length: 0
  180.  
  181.  
  182. <------------>
  183. -- Executing [my-number@incoming:1] Log("SIP/my-number-00000002", "NOTICE,*** Call +41225185576 ***") in new stack
  184. [Feb 14 03:32:02] NOTICE[27551][C-00000002]: Ext. my-number:1 @ incoming: *** Call +41225185576 ***
  185. -- Executing [my-number@incoming:2] Answer("SIP/my-number-00000002", "") in new stack
  186. Audio is at 18406
  187. Adding codec alaw to SDP
  188. Adding non-codec 0x1 (telephone-event) to SDP
  189.  
  190. <--- Reliably Transmitting (no NAT) to 194.192.15.56:5060 --->
  191. SIP/2.0 200 OK
  192. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995421;received=194.192.15.56
  193. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  194. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  195. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  196. CSeq: 1 INVITE
  197. Server: Asterisk PBX 13.13.1
  198. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  199. Supported: replaces, timer
  200. Contact: <sip:my-number@my-ip-address:5060>
  201. Content-Type: application/sdp
  202. Content-Length: 240
  203.  
  204. v=0
  205. o=root 1151749796 1151749796 IN IP4 my-ip-address
  206. s=Asterisk PBX 13.13.1
  207. c=IN IP4 my-ip-address
  208. t=0 0
  209. m=audio 18406 RTP/AVP 8 101
  210. a=rtpmap:8 PCMA/8000
  211. a=rtpmap:101 telephone-event/8000
  212. a=fmtp:101 0-16
  213. a=maxptime:150
  214. a=sendrecv
  215.  
  216. <------------>
  217.  
  218. <--- SIP read from UDP:194.192.15.56:5060 --->
  219. INVITE sip:my-number@my-ip-address:5060 SIP/2.0
  220. CSeq: 1 INVITE
  221. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421
  222. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  223. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  224. To: <sip:my-number@my-ip-address:5060>
  225. Contact: <sip:194.192.15.56:5060;transport=udp>
  226. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
  227. Content-Type: application/sdp
  228. Content-Length: 209
  229. Max-Forwards: 70
  230.  
  231. v=0
  232. o=- 2293553865 2224995421 IN IP4 194.192.15.56
  233. s=VoipSIP
  234. c=IN IP4 194.192.15.56
  235. t=0 0
  236. m=audio 6458 RTP/AVP 8 101
  237. a=rtpmap:8 PCMA/8000
  238. a=rtpmap:101 telephone-event/8000
  239. a=fmtp:101 0-15
  240. a=sendrecv
  241. <------------->
  242. --- (11 headers 10 lines) ---
  243. Sending to 194.192.15.56:5060 (no NAT)
  244. Sending to 194.192.15.56:5060 (no NAT)
  245. Using INVITE request as basis request - 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  246. Found peer 'my-number' for '+41225185576' from 194.192.15.56:5060
  247. == Using SIP RTP CoS mark 5
  248. Found RTP audio format 8
  249. Found RTP audio format 101
  250. Found audio description format PCMA for ID 8
  251. Found audio description format telephone-event for ID 101
  252. Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  253. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  254. Peer audio RTP is at port 194.192.15.56:6458
  255. Looking for my-number in incoming (domain my-ip-address)
  256. sip_route_dump: route/path hop: <sip:194.192.15.56:5060;transport=udp>
  257.  
  258. <--- Transmitting (no NAT) to 194.192.15.56:5060 --->
  259. SIP/2.0 100 Trying
  260. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421;received=194.192.15.56
  261. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  262. To: <sip:my-number@my-ip-address:5060>
  263. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  264. CSeq: 1 INVITE
  265. Server: Asterisk PBX 13.13.1
  266. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  267. Supported: replaces, timer
  268. Contact: <sip:my-number@my-ip-address:5060>
  269. Content-Length: 0
  270.  
  271.  
  272. <------------>
  273. -- Executing [my-number@incoming:1] Log("SIP/my-number-00000003", "NOTICE,*** Call +41225185576 ***") in new stack
  274. [Feb 14 03:32:02] NOTICE[27552][C-00000003]: Ext. my-number:1 @ incoming: *** Call +41225185576 ***
  275. -- Executing [my-number@incoming:2] Answer("SIP/my-number-00000003", "") in new stack
  276. Audio is at 16202
  277. Adding codec alaw to SDP
  278. Adding non-codec 0x1 (telephone-event) to SDP
  279.  
  280. <--- Reliably Transmitting (no NAT) to 194.192.15.56:5060 --->
  281. SIP/2.0 200 OK
  282. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421;received=194.192.15.56
  283. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  284. To: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  285. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  286. CSeq: 1 INVITE
  287. Server: Asterisk PBX 13.13.1
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  289. Supported: replaces, timer
  290. Contact: <sip:my-number@my-ip-address:5060>
  291. Content-Type: application/sdp
  292. Content-Length: 238
  293.  
  294. v=0
  295. o=root 843246073 843246073 IN IP4 my-ip-address
  296. s=Asterisk PBX 13.13.1
  297. c=IN IP4 my-ip-address
  298. t=0 0
  299. m=audio 16202 RTP/AVP 8 101
  300. a=rtpmap:8 PCMA/8000
  301. a=rtpmap:101 telephone-event/8000
  302. a=fmtp:101 0-16
  303. a=maxptime:150
  304. a=sendrecv
  305.  
  306. <------------>
  307. Retransmitting #1 (no NAT) to 194.192.15.56:5060:
  308. SIP/2.0 200 OK
  309. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995421;received=194.192.15.56
  310. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  311. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  312. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  313. CSeq: 1 INVITE
  314. Server: Asterisk PBX 13.13.1
  315. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  316. Supported: replaces, timer
  317. Contact: <sip:my-number@my-ip-address:5060>
  318. Content-Type: application/sdp
  319. Content-Length: 240
  320.  
  321. v=0
  322. o=root 1151749796 1151749796 IN IP4 my-ip-address
  323. s=Asterisk PBX 13.13.1
  324. c=IN IP4 my-ip-address
  325. t=0 0
  326. m=audio 18406 RTP/AVP 8 101
  327. a=rtpmap:8 PCMA/8000
  328. a=rtpmap:101 telephone-event/8000
  329. a=fmtp:101 0-16
  330. a=maxptime:150
  331. a=sendrecv
  332.  
  333. ---
  334. Retransmitting #1 (no NAT) to 194.192.15.56:5060:
  335. SIP/2.0 200 OK
  336. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421;received=194.192.15.56
  337. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  338. To: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  339. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  340. CSeq: 1 INVITE
  341. Server: Asterisk PBX 13.13.1
  342. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  343. Supported: replaces, timer
  344. Contact: <sip:my-number@my-ip-address:5060>
  345. Content-Type: application/sdp
  346. Content-Length: 238
  347.  
  348. v=0
  349. o=root 843246073 843246073 IN IP4 my-ip-address
  350. s=Asterisk PBX 13.13.1
  351. c=IN IP4 my-ip-address
  352. t=0 0
  353. m=audio 16202 RTP/AVP 8 101
  354. a=rtpmap:8 PCMA/8000
  355. a=rtpmap:101 telephone-event/8000
  356. a=fmtp:101 0-16
  357. a=maxptime:150
  358. a=sendrecv
  359.  
  360. ---
  361.  
  362. <--- SIP read from UDP:194.192.15.56:5060 --->
  363. CANCEL sip:my-number@my-ip-address:5060 SIP/2.0
  364. CSeq: 1 CANCEL
  365. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421
  366. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  367. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  368. To: <sip:my-number@my-ip-address:5060>
  369. Contact: <sip:194.192.15.56:5060;transport=udp>
  370. Content-Length: 0
  371. Max-Forwards: 70
  372.  
  373. <------------->
  374. --- (9 headers 0 lines) ---
  375. Sending to 194.192.15.56:5060 (no NAT)
  376.  
  377. <--- Transmitting (no NAT) to 194.192.15.56:5060 --->
  378. SIP/2.0 200 OK
  379. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959551202442224995421;received=194.192.15.56
  380. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  381. To: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  382. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  383. CSeq: 1 CANCEL
  384. Server: Asterisk PBX 13.13.1
  385. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  386. Supported: replaces, timer
  387. Content-Length: 0
  388.  
  389.  
  390. <------------>
  391.  
  392. <--- SIP read from UDP:194.192.15.56:5060 --->
  393. BYE sip:my-number@my-ip-address:5060 SIP/2.0
  394. CSeq: 2 BYE
  395. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995546
  396. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  397. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  398. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  399. Contact: <sip:194.192.15.56:5060;transport=udp>
  400. Content-Length: 0
  401. Max-Forwards: 70
  402.  
  403. <------------->
  404. --- (9 headers 0 lines) ---
  405.  
  406. <--- Reliably Transmitting (no NAT) to 194.192.15.56:5060 --->
  407. SIP/2.0 487 Request Terminated
  408. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995421;received=194.192.15.56
  409. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  410. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  411. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  412. CSeq: 1 INVITE
  413. Server: Asterisk PBX 13.13.1
  414. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  415. Supported: replaces, timer
  416. Content-Length: 0
  417.  
  418.  
  419. <------------>
  420. Sending to 194.192.15.56:5060 (no NAT)
  421. Scheduling destruction of SIP dialog '00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56' in 7232 ms (Method: BYE)
  422.  
  423. <--- Transmitting (no NAT) to 194.192.15.56:5060 --->
  424. SIP/2.0 200 OK
  425. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995546;received=194.192.15.56
  426. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  427. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  428. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  429. CSeq: 2 BYE
  430. Server: Asterisk PBX 13.13.1
  431. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  432. Supported: replaces, timer
  433. Content-Length: 0
  434.  
  435.  
  436. <------------>
  437. == Spawn extension (incoming, my-number, 2) exited non-zero on 'SIP/my-number-00000002'
  438. > 0x7fed500129b0 -- Probation passed - setting RTP source address to 194.192.15.56:6458
  439. -- Executing [my-number@incoming:3] Wait("SIP/my-number-00000003", "2") in new stack
  440.  
  441. <--- SIP read from UDP:194.192.15.56:5060 --->
  442. ACK sip:my-number@my-ip-address:5060 SIP/2.0
  443. CSeq: 1 ACK
  444. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959627188362224995609
  445. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  446. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  447. To: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  448. Contact: <sip:194.192.15.56:5060;transport=udp>
  449. Content-Length: 0
  450. Max-Forwards: 70
  451.  
  452. <------------->
  453. --- (9 headers 0 lines) ---
  454.  
  455. <--- SIP read from UDP:194.192.15.56:5060 --->
  456. ACK sip:my-number@my-ip-address:5060 SIP/2.0
  457. CSeq: 1 ACK
  458. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959627188362224995609
  459. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  460. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  461. To: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  462. Contact: <sip:194.192.15.56:5060;transport=udp>
  463. Content-Length: 0
  464. Max-Forwards: 70
  465.  
  466. <------------->
  467. --- (9 headers 0 lines) ---
  468.  
  469. <--- SIP read from UDP:194.192.15.56:5060 --->
  470. ACK sip:my-number@my-ip-address:5060 SIP/2.0
  471. CSeq: 2 ACK
  472. Via: SIP/2.0/UDP 194.192.15.56:5060;branch=z9hG4bK310959519696682224995546
  473. From: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  474. Call-ID: 00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56
  475. To: <sip:my-number@my-ip-address:5060>;tag=as29f92b25
  476. Contact: <sip:194.192.15.56:5060;transport=udp>
  477. Content-Length: 0
  478. Max-Forwards: 70
  479.  
  480. <------------->
  481. --- (9 headers 0 lines) ---
  482. -- Executing [my-number@incoming:4] ReceiveFAX("SIP/my-number-00000003", "/tmp/incoming-fax.tiff,df") in new stack
  483. -- Channel 'SIP/my-number-00000003' receiving FAX '/tmp/incoming-fax.tiff'
  484. == Using UDPTL CoS mark 5
  485. set_destination: Parsing <sip:194.192.15.56:5060;transport=udp> for address/port to send to
  486. set_destination: set destination to 194.192.15.56:5060
  487. Reliably Transmitting (no NAT) to 194.192.15.56:5060:
  488. INVITE sip:194.192.15.56:5060;transport=udp SIP/2.0
  489. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK0d7e4eec
  490. Max-Forwards: 70
  491. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  492. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  493. Contact: <sip:my-number@my-ip-address:5060>
  494. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  495. CSeq: 102 INVITE
  496. User-Agent: Asterisk PBX 13.13.1
  497. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  498. Supported: replaces, timer
  499. X-asterisk-Info: SIP re-invite (External RTP bridge)
  500. Content-Type: application/sdp
  501. Content-Length: 270
  502.  
  503. v=0
  504. o=root 843246073 843246074 IN IP4 my-ip-address
  505. s=Asterisk PBX 13.13.1
  506. c=IN IP4 my-ip-address
  507. t=0 0
  508. m=image 4477 udptl t38
  509. a=T38FaxVersion:0
  510. a=T38MaxBitRate:14400
  511. a=T38FaxRateManagement:transferredTCF
  512. a=T38FaxMaxDatagram:1400
  513. a=T38FaxUdpEC:t38UDPRedundancy
  514.  
  515. ---
  516. Retransmitting #1 (no NAT) to 194.192.15.56:5060:
  517. INVITE sip:194.192.15.56:5060;transport=udp SIP/2.0
  518. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK0d7e4eec
  519. Max-Forwards: 70
  520. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  521. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  522. Contact: <sip:my-number@my-ip-address:5060>
  523. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  524. CSeq: 102 INVITE
  525. User-Agent: Asterisk PBX 13.13.1
  526. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  527. Supported: replaces, timer
  528. X-asterisk-Info: SIP re-invite (External RTP bridge)
  529. Content-Type: application/sdp
  530. Content-Length: 270
  531.  
  532. v=0
  533. o=root 843246073 843246074 IN IP4 my-ip-address
  534. s=Asterisk PBX 13.13.1
  535. c=IN IP4 my-ip-address
  536. t=0 0
  537. m=image 4477 udptl t38
  538. a=T38FaxVersion:0
  539. a=T38MaxBitRate:14400
  540. a=T38FaxRateManagement:transferredTCF
  541. a=T38FaxMaxDatagram:1400
  542. a=T38FaxUdpEC:t38UDPRedundancy
  543.  
  544. ---
  545.  
  546. <--- SIP read from UDP:194.192.15.56:5060 --->
  547. SIP/2.0 100 Trying
  548. CSeq: 102 INVITE
  549. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK0d7e4eec
  550. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  551. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  552. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  553. Content-Length: 0
  554.  
  555. <------------->
  556. --- (7 headers 0 lines) ---
  557.  
  558. <--- SIP read from UDP:194.192.15.56:5060 --->
  559. SIP/2.0 200 OK
  560. CSeq: 102 INVITE
  561. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK0d7e4eec
  562. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  563. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  564. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  565. Contact: <sip:194.192.15.56:5060;transport=udp>
  566. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
  567. Content-Type: application/sdp
  568. Content-Length: 487
  569.  
  570. v=0
  571. o=- 2293553865 2224995422 IN IP4 194.192.15.56
  572. s=VoipSIP
  573. c=IN IP4 194.192.15.56
  574. t=0 0
  575. m=image 6458 udptl t38
  576. a=T38FaxVersion:0
  577. a=T38FaxRateManagement:transferredTCF
  578. a=T38FaxMaxDatagram:160
  579. a=T38FaxUdpEC:t38UDPRedundancy
  580. a=sqn:0
  581. a=cdsc: 1 audio RTP/AVP 8 101
  582. a=cdsc: 3 image udptl t38
  583. a=cpar: a=T38FaxVersion:0
  584. a=cpar: a=T38FaxRateManagement:transferredTCF
  585. a=cpar: a=T38FaxMaxDatagram:160
  586. a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
  587. a=X-sqn:0
  588. a=X-cap: 1 image udptl t38
  589. <------------->
  590. --- (10 headers 19 lines) ---
  591.  
  592. <--- SIP read from UDP:194.192.15.56:5060 --->
  593. SIP/2.0 200 OK
  594. CSeq: 102 INVITE
  595. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK0d7e4eec
  596. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  597. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  598. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  599. Contact: <sip:194.192.15.56:5060;transport=udp>
  600. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
  601. Content-Type: application/sdp
  602. Content-Length: 487
  603.  
  604. v=0
  605. o=- 2293553865 2224995422 IN IP4 194.192.15.56
  606. s=VoipSIP
  607. c=IN IP4 194.192.15.56
  608. t=0 0
  609. m=image 6458 udptl t38
  610. a=T38FaxVersion:0
  611. a=T38FaxRateManagement:transferredTCF
  612. a=T38FaxMaxDatagram:160
  613. a=T38FaxUdpEC:t38UDPRedundancy
  614. a=sqn:0
  615. a=cdsc: 1 audio RTP/AVP 8 101
  616. a=cdsc: 3 image udptl t38
  617. a=cpar: a=T38FaxVersion:0
  618. a=cpar: a=T38FaxRateManagement:transferredTCF
  619. a=cpar: a=T38FaxMaxDatagram:160
  620. a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
  621. a=X-sqn:0
  622. a=X-cap: 1 image udptl t38
  623. <------------->
  624. --- (10 headers 19 lines) ---
  625. set_destination: Parsing <sip:194.192.15.56:5060;transport=udp> for address/port to send to
  626. set_destination: set destination to 194.192.15.56:5060
  627. Transmitting (no NAT) to 194.192.15.56:5060:
  628. ACK sip:194.192.15.56:5060;transport=udp SIP/2.0
  629. Via: SIP/2.0/UDP my-ip-address:5060;branch=z9hG4bK6d349b41
  630. Max-Forwards: 70
  631. From: <sip:my-number@my-ip-address:5060>;tag=as500cef67
  632. To: "+41225185576" <sip:+41225185576@194.192.15.56:5060>;tag=0931592224995421
  633. Contact: <sip:my-number@my-ip-address:5060>
  634. Call-ID: 0ad49ebc657a72babb71e7c7a003513d457509f4f1@194.192.15.56
  635. CSeq: 102 ACK
  636. User-Agent: Asterisk PBX 13.13.1
  637. Content-Length: 0
  638.  
  639.  
  640. ---
  641. Really destroying SIP dialog '00549ebc656392f4a60fb5e604516714e36b113793@194.192.15.56' Method: BYE
  642. [Feb 14 03:32:12] WARNING[27552][C-00000003]: res_fax.c:1947 receivefax_t38_init: channel 'SIP/my-number-00000003' timed-out during the T.38 negotiation.
  643. Channel 'SIP/my-number-00000003' fax session '1', [ 000.220059 ], stack sent 10 frames (200 ms) of silence.
  644. Channel 'SIP/my-number-00000003' fax session '1', [ 002.820110 ], stack sent 130 frames (2600 ms) of energy.
  645. Channel 'SIP/my-number-00000003' fax session '1', [ 002.880090 ], stack sent 3 frames (60 ms) of silence.
  646. Channel 'SIP/my-number-00000003' fax session '1', [ 004.960089 ], stack sent 104 frames (2080 ms) of energy.
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