API tools faq
paste
Advertisement
Guest User

Untitled

a guest
Mar 8th, 2013
181
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 26.73 KB | None | 0 0
raw download clone embed print report
  1. * Name : Trunk
  2. Secret : <Not set>
  3. MD5Secret : <Not set>
  4. Remote Secret: <Not set>
  5. Context : from-trunk-sip-Trunk
  6. Subscr.Cont. : <Not set>
  7. Language :
  8. AMA flags : Unknown
  9. Transfer mode: open
  10. CallingPres : Presentation Allowed, Not Screened
  11. Callgroup :
  12. Pickupgroup :
  13. MOH Suggest :
  14. Mailbox :
  15. VM Extension : *97
  16. LastMsgsSent : 32767/65535
  17. Call limit : 0
  18. Max forwards : 0
  19. Dynamic : No
  20. Callerid : "" <>
  21. MaxCallBR : 384 kbps
  22. Expire : -1
  23. Insecure : invite
  24. Force rport : No
  25. ACL : No
  26. DirectMedACL : No
  27. T.38 support : No
  28. T.38 EC mode : Unknown
  29. T.38 MaxDtgrm: -1
  30. DirectMedia : Yes
  31. PromiscRedir : No
  32. User=Phone : No
  33. Video Support: No
  34. Text Support : No
  35. Ign SDP ver : No
  36. Trust RPID : No
  37. Send RPID : No
  38. Subscriptions: Yes
  39. Overlap dial : Yes
  40. DTMFmode : rfc2833
  41. Timer T1 : 500
  42. Timer B : 32000
  43. ToHost : 10.100.210.254
  44. Addr->IP : 10.100.210.254:5060
  45. Defaddr->IP : (null)
  46. Prim.Transp. : UDP
  47. Allowed.Trsp : UDP
  48. Def. Username:
  49. SIP Options : (none)
  50. Codecs : 0x10c (ulaw|alaw|g729)
  51. Codec Order : (ulaw:20,alaw:20,g729:40)
  52. Auto-Framing : No
  53. Status : Unmonitored
  54. Useragent :
  55. Reg. Contact :
  56. Qualify Freq : 60000 ms
  57. Sess-Timers : Accept
  58. Sess-Refresh : uas
  59. Sess-Expires : 1800 secs
  60. Min-Sess : 90 secs
  61. RTP Engine : asterisk
  62. Parkinglot :
  63. Use Reason : No
  64. Encryption : No
  65. DND : No
  66. CallFwd Num. :
  67.  
  68.  
  69.  
  70. * Name : 3217
  71. Secret : <Set>
  72. MD5Secret : <Not set>
  73. Remote Secret: <Not set>
  74. Context : forall
  75. Subscr.Cont. : <Not set>
  76. Language :
  77. AMA flags : Unknown
  78. Transfer mode: open
  79. CallingPres : Presentation Allowed, Not Screened
  80. Callgroup :
  81. Pickupgroup :
  82. MOH Suggest :
  83. Mailbox : 3217@device
  84. VM Extension : *97
  85. LastMsgsSent : 0/0
  86. Call limit : 2147483647
  87. Max forwards : 0
  88. Dynamic : Yes
  89. Callerid : "Test" <3217>
  90. MaxCallBR : 384 kbps
  91. Expire : 3192
  92. Insecure : no
  93. Force rport : No
  94. ACL : Yes
  95. DirectMedACL : No
  96. T.38 support : No
  97. T.38 EC mode : Unknown
  98. T.38 MaxDtgrm: -1
  99. DirectMedia : Yes
  100. PromiscRedir : No
  101. User=Phone : No
  102. Video Support: No
  103. Text Support : No
  104. Ign SDP ver : No
  105. Trust RPID : Yes
  106. Send RPID : Yes
  107. Subscriptions: Yes
  108. Overlap dial : Yes
  109. DTMFmode : rfc2833
  110. Timer T1 : 500
  111. Timer B : 32000
  112. ToHost :
  113. Addr->IP : 10.200.210.51:6491
  114. Defaddr->IP : (null)
  115. Prim.Transp. : UDP
  116. Allowed.Trsp : UDP
  117. Def. Username: 3217
  118. SIP Options : (none)
  119. Codecs : 0x100 (g729)
  120. Codec Order : (g729:40)
  121. Auto-Framing : No
  122. Status : OK (388 ms)
  123. Useragent : eyeBeam release 3006o stamp 17551
  124. Reg. Contact : sip:[email protected]:6491
  125. Qualify Freq : 60000 ms
  126. Sess-Timers : Accept
  127. Sess-Refresh : uas
  128. Sess-Expires : 1800 secs
  129. Min-Sess : 90 secs
  130. RTP Engine : asterisk
  131. Parkinglot :
  132. Use Reason : No
  133. Encryption : No
  134. DND : No
  135. CallFwd Num. :
  136.  
  137.  
  138.  
  139. <--- SIP read from UDP:10.200.210.51:6491 --->
  140.  
  141. <------------->
  142. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  143.  
  144. <--- SIP read from UDP:10.200.210.51:6491 --->
  145. INVITE sip:[email protected] SIP/2.0
  146. To: <sip:[email protected]>
  147. From: h<sip:[email protected]>;tag=ef31bf1d
  148. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;rport
  149. Call-ID: f15355682030a66d
  150. CSeq: 1 INVITE
  151. Contact: <sip:[email protected]:6491>
  152. Max-Forwards: 70
  153. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  154. Content-Type: application/sdp
  155. User-Agent: eyeBeam release 3006o stamp 17551
  156. Content-Length: 240
  157.  
  158. v=0
  159. o=- 636137314 636137321 IN IP4 10.200.210.51
  160. s=eyeBeam
  161. c=IN IP4 10.200.210.51
  162. t=0 0
  163. m=audio 6038 RTP/AVP 18 0 8 101
  164. a=alt:1 1 : FCC51B9B 000000FD 10.200.210.51 6038
  165. a=fmtp:101 0-15
  166. a=rtpmap:101 telephone-event/8000
  167. a=sendrecv
  168. <------------->
  169. --- (12 headers 10 lines) ---
  170. Sending to 10.200.210.51:6491 (NAT)
  171. Using INVITE request as basis request - f15355682030a66d
  172. Found peer '3217' for '3217' from 10.200.210.51:6491
  173.  
  174. <--- Reliably Transmitting (no NAT) to 10.200.210.51:6491 --->
  175. SIP/2.0 401 Unauthorized
  176. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;received=10.200.210.51;rport=6491
  177. From: h<sip:[email protected]>;tag=ef31bf1d
  178. To: <sip:[email protected]>;tag=as415b40b8
  179. Call-ID: f15355682030a66d
  180. CSeq: 1 INVITE
  181. Server: Asterisk
  182. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  183. Supported: replaces, timer
  184. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dff0cbb"
  185. Content-Length: 0
  186.  
  187.  
  188. <------------>
  189. Scheduling destruction of SIP dialog 'f15355682030a66d' in 25152 ms (Method: INVITE)
  190.  
  191. <--- SIP read from UDP:10.200.210.51:6491 --->
  192. ACK sip:[email protected] SIP/2.0
  193. To: <sip:[email protected]>;tag=as415b40b8
  194. From: h<sip:[email protected]>;tag=ef31bf1d
  195. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;rport
  196. Call-ID: f15355682030a66d
  197. CSeq: 1 ACK
  198. Content-Length: 0
  199.  
  200. <------------->
  201. --- (7 headers 0 lines) ---
  202.  
  203. <--- SIP read from UDP:10.200.210.51:6491 --->
  204. INVITE sip:[email protected] SIP/2.0
  205. To: <sip:[email protected]>
  206. From: h<sip:[email protected]>;tag=ef31bf1d
  207. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;rport
  208. Call-ID: f15355682030a66d
  209. CSeq: 2 INVITE
  210. Contact: <sip:[email protected]:6491>
  211. Max-Forwards: 70
  212. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  213. Content-Type: application/sdp
  214. User-Agent: eyeBeam release 3006o stamp 17551
  215. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:[email protected]",response="5426822667bb878569ee583c80f153bc",algorithm=MD5
  216. Content-Length: 240
  217.  
  218. v=0
  219. o=- 636137314 636137321 IN IP4 10.200.210.51
  220. s=eyeBeam
  221. c=IN IP4 10.200.210.51
  222. t=0 0
  223. m=audio 6038 RTP/AVP 18 0 8 101
  224. a=alt:1 1 : FCC51B9B 000000FD 10.200.210.51 6038
  225. a=fmtp:101 0-15
  226. a=rtpmap:101 telephone-event/8000
  227. a=sendrecv
  228. <------------->
  229. --- (13 headers 10 lines) ---
  230. Sending to 10.200.210.51:6491 (no NAT)
  231. Using INVITE request as basis request - f15355682030a66d
  232. Found peer '3217' for '3217' from 10.200.210.51:6491
  233. == Using SIP RTP TOS bits 184
  234. == Using SIP RTP CoS mark 5
  235. Found RTP audio format 18
  236. Found RTP audio format 0
  237. Found RTP audio format 8
  238. Found RTP audio format 101
  239. Found audio description format telephone-event for ID 101
  240. Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  241. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  242. Peer audio RTP is at port 10.200.210.51:6038
  243. Looking for 9050555555 in forall (domain 172.20.255.50)
  244. list_route: hop: <sip:[email protected]:6491>
  245.  
  246. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  247. SIP/2.0 100 Trying
  248. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  249. From: h<sip:[email protected]>;tag=ef31bf1d
  250. To: <sip:[email protected]>
  251. Call-ID: f15355682030a66d
  252. CSeq: 2 INVITE
  253. Server: Asterisk
  254. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  255. Supported: replaces, timer
  256. Contact: <sip:[email protected]:5060>
  257. Content-Length: 0
  258.  
  259.  
  260. <------------>
  261. -- Executing [9050555555@forall:1] Macro("SIP/3217-0000000c", "user-callerid,LIMIT,") in new stack
  262. -- Executing [s@macro-user-callerid:1] Set("SIP/3217-0000000c", "AMPUSER=3217") in new stack
  263. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/3217-0000000c", "0?report") in new stack
  264. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/3217-0000000c", "1?Set(REALCALLERIDNUM=3217)") in new stack
  265. -- Executing [s@macro-user-callerid:4] Set("SIP/3217-0000000c", "AMPUSER=3217") in new stack
  266. -- Executing [s@macro-user-callerid:5] Set("SIP/3217-0000000c", "AMPUSERCIDNAME=Test") in new stack
  267. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/3217-0000000c", "0?report") in new stack
  268. -- Executing [s@macro-user-callerid:7] Set("SIP/3217-0000000c", "AMPUSERCID=3217") in new stack
  269. -- Executing [s@macro-user-callerid:8] Set("SIP/3217-0000000c", "CALLERID(all)="Test" <3217>") in new stack
  270. -- Executing [s@macro-user-callerid:9] GotoIf("SIP/3217-0000000c", "0?limit") in new stack
  271. -- Executing [s@macro-user-callerid:10] ExecIf("SIP/3217-0000000c", "1?Set(GROUP(concurrency_limit)=3217)") in new stack
  272. -- Executing [s@macro-user-callerid:11] GosubIf("SIP/3217-0000000c", "7?sub-ccss,s,1(forall,9050555555)") in new stack
  273. -- Executing [s@sub-ccss:1] ExecIf("SIP/3217-0000000c", "0?Return()") in new stack
  274. -- Executing [s@sub-ccss:2] Set("SIP/3217-0000000c", "CCSS_SETUP=TRUE") in new stack
  275. -- Executing [s@sub-ccss:3] GosubIf("SIP/3217-0000000c", "0?monitor_config,1(forall,9050555555):monitor_default,1(forall,9050555555)") in new stack
  276. -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/3217-0000000c", "0?is_exten") in new stack
  277. -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/3217-0000000c", "") in new stack
  278. -- Executing [monitor_default@sub-ccss:3] Return("SIP/3217-0000000c", "FALSE") in new stack
  279. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/3217-0000000c", "1?continue") in new stack
  280. -- Goto (macro-user-callerid,s,25)
  281. -- Executing [s@macro-user-callerid:25] Set("SIP/3217-0000000c", "CALLERID(number)=3217") in new stack
  282. -- Executing [s@macro-user-callerid:26] Set("SIP/3217-0000000c", "CALLERID(name)=Test") in new stack
  283. -- Executing [s@macro-user-callerid:27] Set("SIP/3217-0000000c", "CHANNEL(language)=en") in new stack
  284. -- Executing [9050555555@forall:2] Set("SIP/3217-0000000c", "MOHCLASS=default") in new stack
  285. -- Executing [9050555555@forall:3] Set("SIP/3217-0000000c", "_NODEST=") in new stack
  286. -- Executing [9050555555@forall:4] Gosub("SIP/3217-0000000c", "sub-record-check,s,1(out,9050555555,)") in new stack
  287. -- Executing [s@sub-record-check:1] GotoIf("SIP/3217-0000000c", "1?check") in new stack
  288. -- Goto (sub-record-check,s,6)
  289. -- Executing [s@sub-record-check:6] Set("SIP/3217-0000000c", "__MON_FMT=g729") in new stack
  290. -- Executing [s@sub-record-check:7] GotoIf("SIP/3217-0000000c", "1?next") in new stack
  291. -- Goto (sub-record-check,s,10)
  292. -- Executing [s@sub-record-check:10] ExecIf("SIP/3217-0000000c", "0?Return()") in new stack
  293. -- Executing [s@sub-record-check:11] GotoIf("SIP/3217-0000000c", "0?out,1") in new stack
  294. -- Executing [s@sub-record-check:12] Set("SIP/3217-0000000c", "__REC_STATUS=INITIALIZED") in new stack
  295. -- Executing [s@sub-record-check:13] ExecIf("SIP/3217-0000000c", "0?Set(__REC_POLICY_MODE=)") in new stack
  296. -- Executing [s@sub-record-check:14] Set("SIP/3217-0000000c", "NOW=1362774299") in new stack
  297. -- Executing [s@sub-record-check:15] Set("SIP/3217-0000000c", "__DAY=08") in new stack
  298. -- Executing [s@sub-record-check:16] Set("SIP/3217-0000000c", "__MONTH=03") in new stack
  299. -- Executing [s@sub-record-check:17] Set("SIP/3217-0000000c", "__YEAR=2013") in new stack
  300. -- Executing [s@sub-record-check:18] Set("SIP/3217-0000000c", "__TIMESTR=20130308-152459") in new stack
  301. -- Executing [s@sub-record-check:19] Set("SIP/3217-0000000c", "__FROMEXTEN=3217") in new stack
  302. -- Executing [s@sub-record-check:20] Set("SIP/3217-0000000c", "__CALLFILENAME=out-9050555555-3217-20130308-152459-1362774299.12") in new stack
  303. -- Executing [s@sub-record-check:21] Goto("SIP/3217-0000000c", "out,1") in new stack
  304. -- Goto (sub-record-check,out,1)
  305. -- Executing [out@sub-record-check:1] ExecIf("SIP/3217-0000000c", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
  306. -- Executing [out@sub-record-check:2] GosubIf("SIP/3217-0000000c", "0?record,1(exten,9050555555,3217)") in new stack
  307. -- Executing [out@sub-record-check:3] Return("SIP/3217-0000000c", "") in new stack
  308. -- Executing [9050555555@forall:5] Macro("SIP/3217-0000000c", "dialout-trunk,3,050555555,") in new stack
  309. -- Executing [s@macro-dialout-trunk:1] Set("SIP/3217-0000000c", "DIAL_TRUNK=3") in new stack
  310. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/3217-0000000c", "0?sub-pincheck,s,1()") in new stack
  311. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/3217-0000000c", "0?disabletrunk,1") in new stack
  312. -- Executing [s@macro-dialout-trunk:4] Set("SIP/3217-0000000c", "DIAL_NUMBER=050555555") in new stack
  313. -- Executing [s@macro-dialout-trunk:5] Set("SIP/3217-0000000c", "DIAL_TRUNK_OPTIONS=r") in new stack
  314. -- Executing [s@macro-dialout-trunk:6] Set("SIP/3217-0000000c", "OUTBOUND_GROUP=OUT_3") in new stack
  315. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/3217-0000000c", "1?nomax") in new stack
  316. -- Goto (macro-dialout-trunk,s,9)
  317. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/3217-0000000c", "0?skipoutcid") in new stack
  318. -- Executing [s@macro-dialout-trunk:10] Set("SIP/3217-0000000c", "DIAL_TRUNK_OPTIONS=") in new stack
  319. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/3217-0000000c", "outbound-callerid,3") in new stack
  320. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/3217-0000000c", "0?Set(CALLERPRES()=)") in new stack
  321. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/3217-0000000c", "0?Set(REALCALLERIDNUM=3217)") in new stack
  322. -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/3217-0000000c", "1?normcid") in new stack
  323. -- Goto (macro-outbound-callerid,s,6)
  324. -- Executing [s@macro-outbound-callerid:6] Set("SIP/3217-0000000c", "USEROUTCID=01111") in new stack
  325. -- Executing [s@macro-outbound-callerid:7] Set("SIP/3217-0000000c", "EMERGENCYCID=") in new stack
  326. -- Executing [s@macro-outbound-callerid:8] Set("SIP/3217-0000000c", "TRUNKOUTCID=") in new stack
  327. -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/3217-0000000c", "1?trunkcid") in new stack
  328. -- Goto (macro-outbound-callerid,s,12)
  329. -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/3217-0000000c", "0?Set(CALLERID(all)=)") in new stack
  330. -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/3217-0000000c", "1?Set(CALLERID(all)=01111)") in new stack
  331. -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/3217-0000000c", "0?Set(CALLERID(all)=)") in new stack
  332. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/3217-0000000c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  333. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/3217-0000000c", "0?sub-flp-3,s,1()") in new stack
  334. -- Executing [s@macro-dialout-trunk:13] Set("SIP/3217-0000000c", "OUTNUM=9050555555") in new stack
  335. -- Executing [s@macro-dialout-trunk:14] Set("SIP/3217-0000000c", "custom=SIP/Trunk") in new stack
  336. -- Executing [s@macro-dialout-trunk:15] Macro("SIP/3217-0000000c", "dialout-trunk-predial-hook,") in new stack
  337. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/3217-0000000c", "") in new stack
  338. -- Executing [s@macro-dialout-trunk:16] GotoIf("SIP/3217-0000000c", "0?bypass,1") in new stack
  339. -- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/3217-0000000c", "1?Set(CONNECTEDLINE(num,i)=050555555)") in new stack
  340. -- Executing [s@macro-dialout-trunk:18] ExecIf("SIP/3217-0000000c", "1?Set(CONNECTEDLINE(name,i)=CID:01111)") in new stack
  341. -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/3217-0000000c", "0?customtrunk") in new stack
  342. -- Executing [s@macro-dialout-trunk:20] Dial("SIP/3217-0000000c", "SIP/Trunk/9050555555,300,") in new stack
  343. == Using SIP RTP TOS bits 184
  344. == Using SIP RTP CoS mark 5
  345. Audio is at 25750
  346. Adding codec 0x100 (g729) to SDP
  347. Adding codec 0x4 (ulaw) to SDP
  348. Adding codec 0x8 (alaw) to SDP
  349. Adding non-codec 0x1 (telephone-event) to SDP
  350. Reliably Transmitting (no NAT) to 10.100.210.254:5060:
  351. INVITE sip:[email protected] SIP/2.0
  352. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  353. Max-Forwards: 70
  354. From: "01111" <sip:[email protected]>;tag=as0c91354b
  355. To: <sip:[email protected]>
  356. Contact: <sip:[email protected]:5060>
  357. Call-ID: [email protected]:5060
  358. CSeq: 101 INVITE
  359. User-Agent: Asterisk
  360. Date: Fri, 08 Mar 2013 20:24:59 GMT
  361. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  362. Supported: replaces, timer
  363. Content-Type: application/sdp
  364. Content-Length: 310
  365.  
  366. v=0
  367. o=root 1739091036 1739091036 IN IP4 172.20.255.50
  368. s=Asterisk PBX 1.8.14.0
  369. c=IN IP4 172.20.255.50
  370. t=0 0
  371. m=audio 25750 RTP/AVP 18 0 8 101
  372. a=rtpmap:18 G729/8000
  373. a=fmtp:18 annexb=no
  374. a=rtpmap:0 PCMU/8000
  375. a=rtpmap:8 PCMA/8000
  376. a=rtpmap:101 telephone-event/8000
  377. a=fmtp:101 0-16
  378. a=ptime:20
  379. a=sendrecv
  380.  
  381. ---
  382. -- Called SIP/Trunk/9050555555
  383.  
  384. <--- SIP read from UDP:10.100.210.254:50510 --->
  385. SIP/2.0 100 Trying
  386. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  387. From: "01111" <sip:[email protected]>;tag=as0c91354b
  388. To: <sip:[email protected]>
  389. Date: Fri, 08 Mar 2013 20:25:00 GMT
  390. Call-ID: [email protected]:5060
  391. CSeq: 101 INVITE
  392. Allow-Events: telephone-event
  393. Server: Cisco-SIPGateway/IOS-12.x
  394. Content-Length: 0
  395.  
  396. <------------->
  397. --- (10 headers 0 lines) ---
  398.  
  399. <--- SIP read from UDP:10.100.210.254:50510 --->
  400. SIP/2.0 183 Session Progress
  401. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  402. From: "01111" <sip:[email protected]>;tag=as0c91354b
  403. To: <sip:[email protected]>;tag=26399E4C-EB0
  404. Date: Fri, 08 Mar 2013 20:25:00 GMT
  405. Call-ID: [email protected]:5060
  406. CSeq: 101 INVITE
  407. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
  408. Allow-Events: telephone-event
  409. Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
  410. Contact: <sip:[email protected]:5060>
  411. Supported: sdp-anat
  412. Server: Cisco-SIPGateway/IOS-12.x
  413. Content-Type: application/sdp
  414. Content-Disposition: session;handling=required
  415. Content-Length: 220
  416.  
  417. v=0
  418. o=CiscoSystemsSIP-GW-UserAgent 3635 7391 IN IP4 10.100.210.254
  419. s=SIP Call
  420. c=IN IP4 10.100.210.254
  421. t=0 0
  422. m=audio 17302 RTP/AVP 18
  423. c=IN IP4 10.100.210.254
  424. a=rtpmap:18 G729/8000
  425. a=fmtp:18 annexb=no
  426. a=ptime:20
  427. <------------->
  428. --- (16 headers 10 lines) ---
  429. list_route: hop: <sip:[email protected]:5060>
  430. Found RTP audio format 18
  431. Found audio description format G729 for ID 18
  432. Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  433. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  434. Peer audio RTP is at port 10.100.210.254:17302
  435. -- SIP/Trunk-0000000d is making progress passing it to SIP/3217-0000000c
  436. Audio is at 16414
  437. Adding codec 0x100 (g729) to SDP
  438. Adding non-codec 0x1 (telephone-event) to SDP
  439.  
  440. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  441. SIP/2.0 183 Session Progress
  442. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  443. From: h<sip:[email protected]>;tag=ef31bf1d
  444. To: <sip:[email protected]>;tag=as447a99fe
  445. Call-ID: f15355682030a66d
  446. CSeq: 2 INVITE
  447. Server: Asterisk
  448. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  449. Supported: replaces, timer
  450. Contact: <sip:[email protected]:5060>
  451. Content-Type: application/sdp
  452. Content-Length: 262
  453.  
  454. v=0
  455. o=root 1943296419 1943296419 IN IP4 172.20.255.50
  456. s=Asterisk PBX 1.8.14.0
  457. c=IN IP4 172.20.255.50
  458. t=0 0
  459. m=audio 16414 RTP/AVP 18 101
  460. a=rtpmap:18 G729/8000
  461. a=fmtp:18 annexb=no
  462. a=rtpmap:101 telephone-event/8000
  463. a=fmtp:101 0-16
  464. a=ptime:40
  465. a=sendrecv
  466.  
  467. <------------>
  468.  
  469. <--- SIP read from UDP:10.100.210.254:50510 --->
  470. SIP/2.0 200 OK
  471. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  472. From: "01111" <sip:[email protected]>;tag=as0c91354b
  473. To: <sip:[email protected]>;tag=26399E4C-EB0
  474. Date: Fri, 08 Mar 2013 20:25:00 GMT
  475. Call-ID: [email protected]:5060
  476. CSeq: 101 INVITE
  477. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
  478. Allow-Events: telephone-event
  479. Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
  480. Contact: <sip:[email protected]:5060>
  481. Supported: replaces
  482. Supported: sdp-anat
  483. Server: Cisco-SIPGateway/IOS-12.x
  484. Supported: timer
  485. Content-Type: application/sdp
  486. Content-Disposition: session;handling=required
  487. Content-Length: 220
  488.  
  489. v=0
  490. o=CiscoSystemsSIP-GW-UserAgent 3635 7391 IN IP4 10.100.210.254
  491. s=SIP Call
  492. c=IN IP4 10.100.210.254
  493. t=0 0
  494. m=audio 17302 RTP/AVP 18
  495. c=IN IP4 10.100.210.254
  496. a=rtpmap:18 G729/8000
  497. a=fmtp:18 annexb=no
  498. a=ptime:20
  499. <------------->
  500. --- (18 headers 10 lines) ---
  501. list_route: hop: <sip:[email protected]:5060>
  502. set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  503. set_destination: set destination to 10.100.210.254:5060
  504. Transmitting (no NAT) to 10.100.210.254:5060:
  505. ACK sip:[email protected]:5060 SIP/2.0
  506. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK46af4eff
  507. Max-Forwards: 70
  508. From: "01111" <sip:[email protected]>;tag=as0c91354b
  509. To: <sip:[email protected]>;tag=26399E4C-EB0
  510. Contact: <sip:[email protected]:5060>
  511. Call-ID: [email protected]:5060
  512. CSeq: 101 ACK
  513. User-Agent: Asterisk
  514. Content-Length: 0
  515.  
  516.  
  517. ---
  518. -- SIP/Trunk-0000000d answered SIP/3217-0000000c
  519. Audio is at 16414
  520. Adding codec 0x100 (g729) to SDP
  521. Adding non-codec 0x1 (telephone-event) to SDP
  522.  
  523. <--- Reliably Transmitting (no NAT) to 10.200.210.51:6491 --->
  524. SIP/2.0 200 OK
  525. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  526. From: h<sip:[email protected]>;tag=ef31bf1d
  527. To: <sip:[email protected]>;tag=as447a99fe
  528. Call-ID: f15355682030a66d
  529. CSeq: 2 INVITE
  530. Server: Asterisk
  531. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  532. Supported: replaces, timer
  533. Contact: <sip:[email protected]:5060>
  534. Remote-Party-ID: "CID:01111" <sip:[email protected]>;party=called;privacy=off;screen=no
  535. Content-Type: application/sdp
  536. Content-Length: 262
  537.  
  538. v=0
  539. o=root 1943296419 1943296420 IN IP4 172.20.255.50
  540. s=Asterisk PBX 1.8.14.0
  541. c=IN IP4 172.20.255.50
  542. t=0 0
  543. m=audio 16414 RTP/AVP 18 101
  544. a=rtpmap:18 G729/8000
  545. a=fmtp:18 annexb=no
  546. a=rtpmap:101 telephone-event/8000
  547. a=fmtp:101 0-16
  548. a=ptime:40
  549. a=sendrecv
  550.  
  551. <------------>
  552.  
  553. <--- SIP read from UDP:10.200.210.51:6491 --->
  554. ACK sip:[email protected]:5060 SIP/2.0
  555. To: <sip:[email protected]>;tag=as447a99fe
  556. From: h<sip:[email protected]>;tag=ef31bf1d
  557. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-322595709-1--d87543-;rport
  558. Call-ID: f15355682030a66d
  559. CSeq: 2 ACK
  560. Contact: <sip:[email protected]:6491>
  561. Max-Forwards: 70
  562. User-Agent: eyeBeam release 3006o stamp 17551
  563. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:[email protected]",response="5426822667bb878569ee583c80f153bc",algorithm=MD5
  564. Content-Length: 0
  565.  
  566. <------------->
  567. --- (11 headers 0 lines) ---
  568.  
  569. <--- SIP read from UDP:10.200.210.51:6491 --->
  570.  
  571. <------------->
  572.  
  573. <--- SIP read from UDP:10.200.210.51:6491 --->
  574. BYE sip:[email protected]:5060 SIP/2.0
  575. To: <sip:[email protected]>;tag=as447a99fe
  576. From: h<sip:[email protected]>;tag=ef31bf1d
  577. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-321263389-1--d87543-;rport
  578. Call-ID: f15355682030a66d
  579. CSeq: 3 BYE
  580. Contact: <sip:[email protected]:6491>
  581. Max-Forwards: 70
  582. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  583. User-Agent: eyeBeam release 3006o stamp 17551
  584. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:[email protected]:5060",response="835a7806ada2978a68f9e7e2f3fd1022",algorithm=MD5
  585. Content-Length: 0
  586.  
  587. <------------->
  588. --- (12 headers 0 lines) ---
  589. Sending to 10.200.210.51:6491 (no NAT)
  590. Scheduling destruction of SIP dialog 'f15355682030a66d' in 25152 ms (Method: BYE)
  591.  
  592. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  593. SIP/2.0 200 OK
  594. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-321263389-1--d87543-;received=10.200.210.51;rport=6491
  595. From: h<sip:[email protected]>;tag=ef31bf1d
  596. To: <sip:[email protected]>;tag=as447a99fe
  597. Call-ID: f15355682030a66d
  598. CSeq: 3 BYE
  599. Server: Asterisk
  600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  601. Supported: replaces, timer
  602. Content-Length: 0
  603.  
  604.  
  605. <------------>
  606. -- Executing [h@macro-dialout-trunk:1] Macro("SIP/3217-0000000c", "hangupcall,") in new stack
  607. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3217-0000000c", "1?theend") in new stack
  608. -- Goto (macro-hangupcall,s,3)
  609. -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3217-0000000c", "0?Set(CDR(recordingfile)=)") in new stack
  610. -- Executing [s@macro-hangupcall:4] Hangup("SIP/3217-0000000c", "") in new stack
  611. == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3217-0000000c' in macro 'hangupcall'
  612. == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3217-0000000c'
  613. Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
  614. set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  615. set_destination: set destination to 10.100.210.254:5060
  616. Reliably Transmitting (no NAT) to 10.100.210.254:5060:
  617. BYE sip:[email protected]:5060 SIP/2.0
  618. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK42b8c1cd
  619. Max-Forwards: 70
  620. From: "01111" <sip:[email protected]>;tag=as0c91354b
  621. To: <sip:[email protected]>;tag=26399E4C-EB0
  622. Call-ID: [email protected]:5060
  623. CSeq: 102 BYE
  624. User-Agent: Asterisk
  625. X-Asterisk-HangupCause: Normal Clearing
  626. X-Asterisk-HangupCauseCode: 16
  627. Content-Length: 0
  628.  
  629.  
  630. ---
  631. == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/3217-0000000c' in macro 'dialout-trunk'
  632. == Spawn extension (forall, 9050555555, 5) exited non-zero on 'SIP/3217-0000000c'
  633.  
  634. <--- SIP read from UDP:10.100.210.254:50510 --->
  635. SIP/2.0 200 OK
  636. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK42b8c1cd
  637. From: "01111" <sip:[email protected]>;tag=as0c91354b
  638. To: <sip:[email protected]>;tag=26399E4C-EB0
  639. Date: Fri, 08 Mar 2013 20:25:09 GMT
  640. Call-ID: [email protected]:5060
  641. Server: Cisco-SIPGateway/IOS-12.x
  642. CSeq: 102 BYE
  643. Reason: Q.850;cause=16
  644. P-RTP-Stat: PS=134,OS=2680,PR=78,OR=3120,PL=0,JI=1,LA=348,DU=6
  645. Content-Length: 0
  646.  
  647. <------------->
  648. --- (11 headers 0 lines) ---
  649. Really destroying SIP dialog '[email protected]:5060' Method: INVITE
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!